Merge branch 'topic/asoc' into for-linus
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 598c22f..5de23c0 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -4288,7 +4288,7 @@
<![CDATA[
struct snd_rawmidi *rmidi;
snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags,
- irq, irq_flags, &rmidi);
+ irq, &rmidi);
]]>
</programlisting>
</informalexample>
@@ -4343,6 +4343,13 @@
by itself to start processing the output stream in the irq handler.
</para>
+ <para>
+ If the MPU-401 interface shares its interrupt with the other logical
+ devices on the card, set <constant>MPU401_INFO_IRQ_HOOK</constant>
+ (see <link linkend="midi-interface-interrupt-handler"><citetitle>
+ below</citetitle></link>).
+ </para>
+
<para>
Usually, the port address corresponds to the command port and
port + 1 corresponds to the data port. If not, you may change
@@ -4375,14 +4382,12 @@
</para>
<para>
- The 6th argument specifies the irq number for UART. If the irq
- is already allocated, pass 0 to the 7th argument
- (<parameter>irq_flags</parameter>). Otherwise, pass the flags
- for irq allocation
- (<constant>SA_XXX</constant> 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.
+ 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.
</para>
</section>
@@ -4390,12 +4395,13 @@
<title>Interrupt Handler</title>
<para>
When the interrupt is allocated in
- <function>snd_mpu401_uart_new()</function>, 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
- <function>snd_mpu401_uart_interrupt()</function> explicitly when
- a UART interrupt is invoked and checked in your own interrupt
- handler.
+ <function>snd_mpu401_uart_new()</function>, 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 <constant>MPU401_INFO_IRQ_HOOK</constant>, and call
+ <function>snd_mpu401_uart_interrupt()</function> explicitly from your
+ own interrupt handler when it has determined that a UART interrupt
+ has occurred.
</para>
<para>
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 8975701..936699e 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -886,6 +886,12 @@
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/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
index 1482035..e9621e3 100644
--- a/Documentation/sound/alsa/HD-Audio-Controls.txt
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -98,3 +98,19 @@
* 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/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index d70c93b..4f34432 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -29,9 +29,6 @@
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)
@@ -46,15 +43,10 @@
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)
@@ -66,43 +58,15 @@
ALC267/268
==========
- 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)
+ N/A
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
@@ -111,15 +75,10 @@
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
======
- base Base model (ASUS NX90)
- auto auto-config reading BIOS (default)
+ N/A
ALC882/883/885/888/889
======================
@@ -175,28 +134,11 @@
ALC861/660
==========
- 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)
+ N/A
ALC861VD/660VD
==============
- 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)
+ N/A
CMI9880
=======
@@ -289,7 +231,6 @@
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/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c82beb0..03e2771 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -447,7 +447,10 @@
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.
+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.
The `[model]` line allows to change the model name of the each codec.
In the example above, it will be changed to model=auto.
@@ -491,7 +494,7 @@
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
@@ -524,6 +527,54 @@
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/MAINTAINERS b/MAINTAINERS
index bbf42cd..1dbbb27 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5991,7 +5991,7 @@
M: Takashi Iwai <tiwai@suse.de>
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-2.6.git
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
T: git git://git.alsa-project.org/alsa-kernel.git
S: Maintained
F: Documentation/sound/
diff --git a/include/linux/input.h b/include/linux/input.h
index a637e78..a514fb8 100644
--- a/include/linux/input.h
+++ b/include/linux/input.h
@@ -814,6 +814,7 @@
#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/include/linux/usb/ch9.h b/include/linux/usb/ch9.h
index 0fd3fbd..f302535 100644
--- a/include/linux/usb/ch9.h
+++ b/include/linux/usb/ch9.h
@@ -377,12 +377,6 @@
#define USB_ENDPOINT_NUMBER_MASK 0x0f /* in bEndpointAddress */
#define USB_ENDPOINT_DIR_MASK 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_XFERTYPE_MASK 0x03 /* in bmAttributes */
#define USB_ENDPOINT_XFER_CONTROL 0
#define USB_ENDPOINT_XFER_ISOC 1
@@ -390,6 +384,17 @@
#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 */
+
/*-------------------------------------------------------------------------*/
/**
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 5d6074f..a2e4ff5 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -706,7 +706,7 @@
* *
****************************************************************************/
-#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6)
+#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7)
struct snd_ctl_card_info {
int card; /* card number */
@@ -803,6 +803,8 @@
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/include/sound/initval.h b/include/sound/initval.h
index 1daa6dff..f99a0d2 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -62,7 +62,7 @@
{
while (*irq_table != -1) {
if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
- IRQF_DISABLED | IRQF_PROBE_SHARED, "ALSA Test IRQ",
+ IRQF_PROBE_SHARED, "ALSA Test IRQ",
(void *) irq_table)) {
free_irq(*irq_table, (void *) irq_table);
return *irq_table;
diff --git a/include/sound/jack.h b/include/sound/jack.h
index c140fc7..63c7907 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -42,6 +42,7 @@
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/include/sound/mpu401.h b/include/sound/mpu401.h
index 1f1d53f..20230db 100644
--- a/include/sound/mpu401.h
+++ b/include/sound/mpu401.h
@@ -50,7 +50,10 @@
#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
@@ -73,8 +76,7 @@
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 (-1 = poll) */
- int irq_flags;
+ int irq; /* IRQ number of MPU-401 chip */
unsigned long mode; /* MPU401_MODE_XXXX */
int timer_invoked;
@@ -131,7 +133,6 @@
unsigned long port,
unsigned int info_flags,
int irq,
- int irq_flags,
struct snd_rawmidi ** rrawmidi);
#endif /* __SOUND_MPU401_H */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 57e71fa..3e7fda6 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -825,6 +825,8 @@
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,
@@ -1035,6 +1037,8 @@
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/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index 3687a6c..762af68 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1067,7 +1067,6 @@
printk(KERN_DEBUG PFX "created and attached onyx instance\n");
return 0;
fail:
- i2c_set_clientdata(client, NULL);
kfree(onyx);
return -ENODEV;
}
@@ -1112,8 +1111,7 @@
aoa_codec_unregister(&onyx->codec);
of_node_put(onyx->codec.node);
- if (onyx->codec_info)
- kfree(onyx->codec_info);
+ kfree(onyx->codec_info);
kfree(onyx);
return 0;
}
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e..552b97a 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@
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/sound/arm/aaci.c b/sound/arm/aaci.c
index d0cead3..e518d38 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -443,7 +443,7 @@
mutex_lock(&aaci->irq_lock);
if (!aaci->users++) {
ret = request_irq(aaci->dev->irq[0], aaci_irq,
- IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci);
+ IRQF_SHARED, DRIVER_NAME, aaci);
if (ret != 0)
aaci->users--;
}
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 88eec38..8ad6535 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -359,7 +359,7 @@
if (ret)
goto err_clk2;
- ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
+ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL);
if (ret < 0)
goto err_irq;
diff --git a/sound/core/control.c b/sound/core/control.c
index f8c5be4..978fe1a 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -989,7 +989,6 @@
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,
@@ -1001,6 +1000,28 @@
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)
{
@@ -1055,11 +1076,46 @@
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;
- if (ue->tlv_data)
- kfree(ue->tlv_data);
+
+ kfree(ue->tlv_data);
+ kfree(ue->priv_data);
kfree(ue);
}
@@ -1072,8 +1128,8 @@
long private_size;
struct user_element *ue;
int idx, err;
-
- if (card->user_ctl_count >= MAX_USER_CONTROLS)
+
+ if (!replace && card->user_ctl_count >= MAX_USER_CONTROLS)
return -ENOMEM;
if (info->count < 1)
return -EINVAL;
@@ -1101,7 +1157,10 @@
memcpy(&kctl.id, &info->id, sizeof(info->id));
kctl.count = info->owner ? info->owner : 1;
access |= SNDRV_CTL_ELEM_ACCESS_USER;
- kctl.info = snd_ctl_elem_user_info;
+ if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED)
+ kctl.info = snd_ctl_elem_user_enum_info;
+ else
+ 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)
@@ -1122,6 +1181,11 @@
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)
@@ -1143,9 +1207,17 @@
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/sound/core/control_compat.c b/sound/core/control_compat.c
index 4268744..2bb95a7 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -83,6 +83,8 @@
u32 items;
u32 item;
char name[64];
+ u64 names_ptr;
+ u32 names_length;
} enumerated;
unsigned char reserved[128];
} value;
@@ -372,6 +374,8 @@
&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/sound/core/jack.c b/sound/core/jack.c
index 53b53e9..240a3e1 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -30,6 +30,7 @@
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/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index d8359cf..1b5e0c4 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -499,7 +499,7 @@
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, name);
+ strlcpy(id.name, name, sizeof(id.name));
id.index = index;
return snd_ctl_find_id(card, &id);
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf..95d1e78 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1399,6 +1399,32 @@
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)
{
@@ -1761,6 +1787,10 @@
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 {
@@ -1771,16 +1801,32 @@
}
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_interruptible(wait_time);
+
+ tout = schedule_timeout(wait_time);
+
snd_pcm_stream_lock_irq(substream);
+ set_current_state(TASK_INTERRUPTIBLE);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
@@ -1806,14 +1852,9 @@
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/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1c6be91..77d7df2 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -2058,16 +2058,12 @@
static int snd_pcm_open_file(struct file *file,
struct snd_pcm *pcm,
- int stream,
- struct snd_pcm_file **rpcm_file)
+ int stream)
{
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;
@@ -2083,8 +2079,7 @@
substream->pcm_release = pcm_release_private;
}
file->private_data = pcm_file;
- if (rpcm_file)
- *rpcm_file = pcm_file;
+
return 0;
}
@@ -2113,7 +2108,6 @@
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) {
@@ -2131,7 +2125,7 @@
add_wait_queue(&pcm->open_wait, &wait);
mutex_lock(&pcm->open_mutex);
while (1) {
- err = snd_pcm_open_file(file, pcm, stream, &pcm_file);
+ err = snd_pcm_open_file(file, pcm, stream);
if (err >= 0)
break;
if (err == -EAGAIN) {
@@ -3156,8 +3150,8 @@
/*
* mmap the DMA buffer on RAM
*/
-static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *area)
+int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area)
{
area->vm_flags |= VM_RESERVED;
#ifdef ARCH_HAS_DMA_MMAP_COHERENT
@@ -3177,6 +3171,7 @@
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
@@ -3242,7 +3237,7 @@
if (substream->ops->mmap)
err = substream->ops->mmap(substream, area);
else
- err = snd_pcm_default_mmap(substream, area);
+ err = snd_pcm_lib_default_mmap(substream, area);
if (!err)
atomic_inc(&substream->mmap_count);
return err;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0..67ebf1c 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@
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) {
@@ -353,6 +355,7 @@
}
mutex_unlock(®ister_mutex);
}
+ out:
if (timeri->private_free)
timeri->private_free(timeri);
kfree(timeri->owner);
@@ -531,6 +534,8 @@
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/sound/drivers/aloop.c b/sound/drivers/aloop.c
index a0da775..4067f15 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -575,7 +575,8 @@
static int loopback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ return snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(params));
}
static int loopback_hw_free(struct snd_pcm_substream *substream)
@@ -587,7 +588,7 @@
mutex_lock(&dpcm->loopback->cable_lock);
cable->valid &= ~(1 << substream->stream);
mutex_unlock(&dpcm->loopback->cable_lock);
- return snd_pcm_lib_free_pages(substream);
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
}
static unsigned int get_cable_index(struct snd_pcm_substream *substream)
@@ -740,6 +741,8 @@
.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 = {
@@ -751,6 +754,8 @@
.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,
@@ -771,10 +776,6 @@
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/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index 5cfcb90..2c7a763 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@
"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, IRQF_DISABLED,
+ if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
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 @@
"request (playback) irq %d done\n",
ml403_ac97cr->irq);
irq = platform_get_irq(pfdev, 1);
- if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+ if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
"unable to grab IRQ %d\n",
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 149d05a..1c02852 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -86,8 +86,7 @@
}
err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0,
- irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0,
- NULL);
+ irq[dev], NULL);
if (err < 0) {
printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]);
goto _err;
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 2af0999..e91698a 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/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. Also, if irq < 0, then
+ * interrupts thus output is done via polling. Without interrupt,
* input is done also via polling. Do not expect good performance.
*
*
@@ -374,7 +374,7 @@
/* first time - flush FIFO */
while (max-- > 0)
mpu->read(mpu, MPU401D(mpu));
- if (mpu->irq < 0)
+ if (mpu->info_flags & MPU401_INFO_USE_TIMER)
snd_mpu401_uart_add_timer(mpu, 1);
}
@@ -383,7 +383,7 @@
snd_mpu401_uart_input_read(mpu);
spin_unlock_irqrestore(&mpu->input_lock, flags);
} else {
- if (mpu->irq < 0)
+ if (mpu->info_flags & MPU401_INFO_USE_TIMER)
snd_mpu401_uart_remove_timer(mpu, 1);
clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode);
}
@@ -496,7 +496,7 @@
static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
{
struct snd_mpu401 *mpu = rmidi->private_data;
- if (mpu->irq_flags && mpu->irq >= 0)
+ if (mpu->irq >= 0)
free_irq(mpu->irq, (void *) mpu);
release_and_free_resource(mpu->res);
kfree(mpu);
@@ -509,8 +509,7 @@
* @hardware: the hardware type, MPU401_HW_XXXX
* @port: the base address of MPU401 port
* @info_flags: bitflags MPU401_INFO_XXX
- * @irq: the irq number, -1 if no interrupt for mpu
- * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved.
+ * @irq: the ISA irq number, -1 if not to be allocated
* @rrawmidi: the pointer to store the new rawmidi instance
*
* Creates a new MPU-401 instance.
@@ -525,7 +524,7 @@
unsigned short hardware,
unsigned long port,
unsigned int info_flags,
- int irq, int irq_flags,
+ int irq,
struct snd_rawmidi ** rrawmidi)
{
struct snd_mpu401 *mpu;
@@ -577,8 +576,8 @@
mpu->cport = port + 2;
else
mpu->cport = port + 1;
- if (irq >= 0 && irq_flags) {
- if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags,
+ if (irq >= 0) {
+ if (request_irq(irq, snd_mpu401_uart_interrupt, 0,
"MPU401 UART", (void *) mpu)) {
snd_printk(KERN_ERR "mpu401_uart: "
"unable to grab IRQ %d\n", irq);
@@ -586,9 +585,10 @@
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/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5c426df..1eef4cc 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -589,7 +589,7 @@
return -EBUSY;
}
mcard->port = port;
- if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
+ if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) {
snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq);
return -EBUSY;
}
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index a25fb7b..fc1d822 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -816,7 +816,7 @@
if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
if (request_irq(irq, snd_uart16550_interrupt,
- IRQF_DISABLED, "Serial MIDI", uart)) {
+ 0, "Serial MIDI", uart)) {
snd_printk(KERN_WARNING
"irq %d busy. Using Polling.\n", irq);
} else {
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 4400308..cd094ec 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -51,7 +51,6 @@
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/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 3fc257d..cbe6bb9 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -778,9 +778,10 @@
{
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);
@@ -796,8 +797,8 @@
fcp_bus_reset(fwspk->unit);
if (cmp_connection_update(&fwspk->connection) < 0) {
- mutex_lock(&fwspk->mutex);
amdtp_out_stream_pcm_abort(&fwspk->stream);
+ mutex_lock(&fwspk->mutex);
fwspk_stop_stream(fwspk);
mutex_unlock(&fwspk->mutex);
return;
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
index 3cb75bc..a87a2b5 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -204,7 +204,7 @@
if (mpu_port[dev] > 0) {
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED,
+ mpu_port[dev], 0, mpu_irq[dev],
NULL) < 0)
printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]);
}
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index 05aef8b..177eed3 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -595,7 +595,7 @@
snd_ad1816a_free(chip);
return -EBUSY;
}
- if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) {
+ if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) {
snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq);
snd_ad1816a_free(chip);
return -EBUSY;
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 20becc8..706effd 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -256,7 +256,6 @@
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/sound/isa/azt2320.c b/sound/isa/azt2320.c
index aac8dc1..b7bdbf3 100644
--- a/sound/isa/azt2320.c
+++ b/sound/isa/azt2320.c
@@ -234,8 +234,7 @@
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], IRQF_DISABLED,
- NULL) < 0)
+ mpu_irq[dev], NULL) < 0)
snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
}
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index fe79a16..dca69f8 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -597,7 +597,7 @@
if (mpuport[dev] != SNDRV_AUTO_PORT) {
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
mpuport[dev], 0, mpuirq[dev],
- IRQF_DISABLED, NULL) < 0)
+ NULL) < 0)
printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n",
mpuport[dev]);
}
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index cb9153e..409fa0a 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -131,7 +131,6 @@
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/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 999dc1e..0dbde46 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -449,8 +449,7 @@
mpu_irq[dev] = -1;
if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
mpu_port[dev], 0,
- mpu_irq[dev],
- mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0)
+ mpu_irq[dev], NULL) < 0)
printk(KERN_WARNING IDENT ": MPU401 not detected\n");
}
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index 0cde813..5493e9e 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -174,7 +174,7 @@
chip->mpu_port > 0) {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
chip->mpu_port, 0,
- mpu_irq[n], IRQF_DISABLED, NULL);
+ mpu_irq[n], NULL);
if (error < 0)
return error;
}
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 0767620..d3eab6f 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -661,7 +661,7 @@
snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
return -EBUSY;
}
- if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) {
+ if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
return -EBUSY;
}
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index fb4d6b3..bf6ad0b 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1805,7 +1805,7 @@
return -EBUSY;
}
- if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx",
+ if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx",
(void *) card)) {
snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
@@ -2160,8 +2160,8 @@
if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX,
- mpu_port[dev], 0,
- irq[dev], 0, &chip->rmidi);
+ mpu_port[dev], MPU401_INFO_IRQ_HOOK,
+ -1, &chip->rmidi);
if (err < 0)
return err;
}
diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c
index ee54df0..e51d324 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -585,8 +585,7 @@
if (mpu_port[n] >= 0) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port[n], 0, mpu_irq[n],
- IRQF_DISABLED, NULL);
+ mpu_port[n], 0, mpu_irq[n], NULL);
if (err < 0)
goto error;
}
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 12eb98f..3167e5a 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -180,7 +180,7 @@
snd_gus_free(gus);
return -EBUSY;
}
- if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) {
+ if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "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/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index 008e8e5..c4733c0 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -317,8 +317,7 @@
if (es1688->mpu_port >= 0x300) {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
- es1688->mpu_port, 0,
- mpu_irq[n], IRQF_DISABLED, NULL);
+ es1688->mpu_port, 0, mpu_irq[n], NULL);
if (error < 0)
goto out;
}
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index 3e4a58b..c43faa0 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -291,7 +291,7 @@
goto _err;
}
- if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) {
+ if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) {
snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
err = -EBUSY;
goto _err;
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index c7b80e4..5f869a3 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -684,7 +684,7 @@
if ((err = snd_gus_initialize(gus)) < 0)
return err;
- if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED,
+ if (request_irq(xirq, snd_interwave_interrupt, 0,
"InterWave", iwcard)) {
snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
return -EBUSY;
diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c
index 91d6023..0961e2c 100644
--- a/sound/isa/msnd/msnd_pinnacle.c
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -600,7 +600,7 @@
mpu_io[0],
MPU401_MODE_INPUT |
MPU401_MODE_OUTPUT,
- mpu_irq[0], IRQF_DISABLED,
+ mpu_irq[0],
&chip->rmidi);
if (err < 0) {
printk(KERN_ERR LOGNAME
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 9b915e2..bbafb0b 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -667,7 +667,7 @@
err = snd_opl3sa2_detect(card);
if (err < 0)
return err;
- err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED,
+ err = request_irq(xirq, snd_opl3sa2_interrupt, 0,
"OPL3-SA2", card);
if (err) {
snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq);
@@ -707,8 +707,9 @@
}
if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2,
- midi_port[dev], 0,
- xirq, 0, &chip->rmidi)) < 0)
+ midi_port[dev],
+ MPU401_INFO_IRQ_HOOK, -1,
+ &chip->rmidi)) < 0)
return err;
}
sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 8c24102..d94d0f3 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -1377,8 +1377,7 @@
rmidi = NULL;
else {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port, 0, miro->mpu_irq, IRQF_DISABLED,
- &rmidi);
+ mpu_port, 0, miro->mpu_irq, &rmidi);
if (error < 0)
snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
mpu_port);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index c35dc68..6dbbfa7 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -892,7 +892,7 @@
#endif
#ifdef OPTi93X
error = request_irq(irq, snd_opti93x_interrupt,
- IRQF_DISABLED, DEV_NAME" - WSS", chip);
+ 0, 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 @@
rmidi = NULL;
else {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi);
+ mpu_port, 0, mpu_irq, &rmidi);
if (error)
snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
mpu_port);
diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c
index 8ccbcdd..54e3c2c 100644
--- a/sound/isa/sb/jazz16.c
+++ b/sound/isa/sb/jazz16.c
@@ -322,7 +322,6 @@
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/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
index 4d1c5a3..237f8bd 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -394,8 +394,9 @@
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, 0,
- xirq, 0, &chip->rmidi)) < 0)
+ chip->mpu_port,
+ MPU401_INFO_IRQ_HOOK, -1,
+ &chip->rmidi)) < 0)
return err;
chip->rmidi_callback = snd_mpu401_uart_interrupt;
}
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index eae6c1c..d2e1921 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -240,7 +240,7 @@
if (request_irq(irq, irq_handler,
(hardware == SB_HW_ALS4000 ||
hardware == SB_HW_CS5530) ?
- IRQF_SHARED : IRQF_DISABLED,
+ IRQF_SHARED : 0,
"SoundBlaster", (void *) chip)) {
snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
snd_sbdsp_free(chip);
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index 9a8bbf6..207c161 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -658,8 +658,7 @@
if (snd_mpu401_uart_new(card, 0,
MPU401_HW_MPU401,
mpu_port[dev], 0,
- mpu_irq[dev], IRQF_DISABLED,
- NULL) < 0)
+ mpu_irq[dev], NULL) < 0)
snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n",
mpu_port[dev]);
}
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index e2d5d2d..f2379e1 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -825,8 +825,7 @@
int err;
err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port,
- MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED,
- &rawmidi);
+ MPU401_INFO_INTEGRATED, irq, &rawmidi);
if (err == 0) {
struct snd_mpu401 *mpu = rawmidi->private_data;
mpu->open_input = mpu401_open;
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
index 711670e..8714297 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -418,7 +418,7 @@
return -EBUSY;
}
if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt,
- IRQF_DISABLED, "ICS2115", acard)) {
+ 0, "ICS2115", acard)) {
snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]);
return -EBUSY;
}
@@ -449,8 +449,7 @@
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], IRQF_DISABLED,
- NULL);
+ cs4232_mpu_irq[dev], NULL);
if (err < 0) {
snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n");
return err;
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 2a42cc3..7277c5b 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1833,7 +1833,7 @@
}
chip->cport = cport;
if (!(hwshare & WSS_HWSHARE_IRQ))
- if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED,
+ if (request_irq(irq, snd_wss_interrupt, 0,
"WSS", (void *) chip)) {
snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq);
snd_wss_free(chip);
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 446cf97..7567ebd 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -465,13 +465,13 @@
flags = claim_dma_lock();
if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
- "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED,
+ "AC97 TX", au1000_dma_interrupt, 0,
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, IRQF_DISABLED,
+ "AC97 RX", au1000_dma_interrupt, 0,
au1000->stream[CAPTURE])) < 0){
release_dma_lock(flags);
return -EBUSY;
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 8f7d175..6f13ab4 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@
if (pcm_channels & 2)
{
- foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
- arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+ foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
+ arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
}
else
{
- foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
- arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+ foo = (PIT_TICK_RATE + (arg / 2)) / arg;
+ arg = (PIT_TICK_RATE + (foo / 2)) / foo;
}
pcm_speed = arg;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 9b800ce..2fc0624 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@
if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */
printk(KERN_INFO "PSS: CDROM port not enabled.\n");
- } else if (check_region(pss_cdrom_port, 2)) {
+ } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
+ pss_cdrom_port = -1;
printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
} else {
set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1232,7 +1233,8 @@
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/sound/oss/sound_timer.c b/sound/oss/sound_timer.c
index 48cda6c..8021c85 100644
--- a/sound/oss/sound_timer.c
+++ b/sound/oss/sound_timer.c
@@ -320,7 +320,7 @@
n = sound_alloc_timerdev();
if (n == -1)
n = 0; /* Overwrite the system timer */
- strcpy(sound_timer.info.name, name);
+ strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name));
sound_timer_devs[n] = &sound_timer;
}
EXPORT_SYMBOL(sound_timer_init);
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5b..8816804 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
# 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
@@ -563,11 +568,6 @@
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/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@
0x103c0944, /* HP nc6220 */
0x103c0934, /* HP nc8220 */
0x103c006d, /* HP nx9105 */
+ 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
0x17340088, /* FSC Scenic-W */
0 /* end */
};
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index a9c1af3..0462869 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -931,8 +931,9 @@
if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000,
iobase + ALS4K_IOB_30_MIDI_DATA,
- MPU401_INFO_INTEGRATED,
- pci->irq, 0, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &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/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca1..bd47521 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@
if (!p_cache)
return NULL;
- p_cache->p_info =
- kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+ p_cache->p_info = kzalloc(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/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index 0dc8d25..e6c6a0f 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -84,7 +84,7 @@
#ifdef VORTEX_MPU401_LEGACY
if ((temp =
snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330,
- 0, 0, 0, &rmidi)) != 0) {
+ MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
hwwrite(vortex->mmio, VORTEX_CTRL,
(hwread(vortex->mmio, VORTEX_CTRL) &
~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
@@ -94,8 +94,8 @@
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,
- 0, 0, &rmidi)) != 0) {
+ MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO |
+ MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
hwwrite(vortex->mmio, VORTEX_CTRL,
(hwread(vortex->mmio, VORTEX_CTRL) &
~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a2..d24fe42 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@
int err;
snd_azf3328_dbgcallenter();
- if (dev >= SNDRV_CARDS)
- return -ENODEV;
+ if (dev >= SNDRV_CARDS) {
+ err = -ENODEV;
+ goto out;
+ }
if (!enable[dev]) {
dev++;
- return -ENOENT;
+ err = -ENOENT;
+ goto out;
}
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
- return err;
+ goto out;
strcpy(card->driver, "AZF3328");
strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
@@ -2649,8 +2652,9 @@
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,
- pci->irq, 0, &chip->rmidi
+ MPU401_HW_AZT2320, chip->mpu_io,
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+ -1, &chip->rmidi
);
if (err < 0) {
snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n",
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 9cf99fb..da9c732 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3228,8 +3228,9 @@
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
iomidi,
(integrated_midi ?
- MPU401_INFO_INTEGRATED : 0),
- cm->irq, 0, &cm->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED : 0) |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &cm->rmidi)) < 0) {
printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
}
}
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 457d211..2c86226 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -404,7 +404,7 @@
int err;
int playback_count, capture_count;
- playback_count = (IEC958 == device) ? 1 : 8;
+ playback_count = (IEC958 == device) ? 1 : 256;
capture_count = (FRONT == device) ? 1 : 0;
err = snd_pcm_new(atc->card, "ctxfi", device,
playback_count, capture_count, &pcm);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index c749fa7..e134b3a 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -20,7 +20,7 @@
#include "cthardware.h"
#include <linux/slab.h>
-#define SRC_RESOURCE_NUM 64
+#define SRC_RESOURCE_NUM 256
#define SRCIMP_RESOURCE_NUM 256
static unsigned int conj_mask;
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
index b23adfc..e6da60e 100644
--- a/sound/pci/ctxfi/ctvmem.h
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -18,7 +18,7 @@
#ifndef CTVMEM_H
#define CTVMEM_H
-#define CT_PTP_NUM 1 /* num of device page table pages */
+#define CT_PTP_NUM 4 /* num of device page table pages */
#include <linux/mutex.h>
#include <linux/list.h>
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 622bace..e22b8e2 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1146,6 +1146,11 @@
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/sound/pci/es1938.c b/sound/pci/es1938.c
index 26a5a2f..718a264 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1854,8 +1854,9 @@
}
}
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- chip->mpu_port, MPU401_INFO_INTEGRATED,
- chip->irq, 0, &chip->rmidi) < 0) {
+ chip->mpu_port,
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+ -1, &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/sound/pci/es1968.c b/sound/pci/es1968.c
index 99ea932..407e4ab 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2843,8 +2843,9 @@
if (enable_mpu[dev]) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
chip->io_port + ESM_MPU401_PORT,
- MPU401_INFO_INTEGRATED,
- chip->irq, 0, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &chip->rmidi)) < 0) {
printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n");
}
}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f0..136f723 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@
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)
@@ -728,11 +729,14 @@
{ .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 = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+ struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
reg &= ~(FM801_GPIO_GP(gpio.data) |
FM801_GPIO_GP(gpio.clk) |
@@ -750,7 +754,7 @@
{
struct fm801 *chip = tea->private_data;
unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
- struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+ struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 |
(reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0;
@@ -760,7 +764,7 @@
{
struct fm801 *chip = tea->private_data;
unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
- struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
+ struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
/* use GPIO lines and set write enable bit */
reg |= FM801_GPIO_GS(gpio.data) |
@@ -1150,7 +1154,8 @@
__end_hw:
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
- snd_tea575x_exit(&chip->tea);
+ if (!(chip->tea575x_tuner & TUNER_DISABLED))
+ snd_tea575x_exit(&chip->tea);
#endif
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -1236,7 +1241,6 @@
(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) {
@@ -1245,17 +1249,19 @@
chip->tea575x_tuner = tea575x_tuner;
if (!snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
- snd_fm801_tea575x_gpios[tea575x_tuner - 1].name);
+ get_tea575x_gpio(chip)->name);
break;
}
}
if (tea575x_tuner == 4) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- snd_fm801_free(chip);
- return -ENODEV;
+ chip->tea575x_tuner = TUNER_DISABLED;
}
}
- strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+ if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+ strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name,
+ sizeof(chip->tea.card));
+ }
#endif
*rchip = chip;
@@ -1306,8 +1312,9 @@
}
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801,
FM801_REG(chip, MPU401_DATA),
- MPU401_INFO_INTEGRATED,
- chip->irq, 0, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &chip->rmidi)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 87365d5..f928d66 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -6,6 +6,9 @@
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/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c
index 21ec2cb..3b5170b 100644
--- a/sound/pci/hda/alc260_quirks.c
+++ b/sound/pci/hda/alc260_quirks.c
@@ -7,9 +7,6 @@
enum {
ALC260_AUTO,
ALC260_BASIC,
- ALC260_HP,
- ALC260_HP_DC7600,
- ALC260_HP_3013,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
@@ -142,8 +139,6 @@
/* 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
*/
@@ -170,145 +165,6 @@
{ } /* 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.
*/
@@ -480,106 +336,6 @@
{ }
};
-#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.
@@ -1093,9 +849,6 @@
*/
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",
@@ -1112,15 +865,6 @@
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),
@@ -1144,54 +888,6 @@
.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/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c
index 8d2097d..7894b2b 100644
--- a/sound/pci/hda/alc262_quirks.c
+++ b/sound/pci/hda/alc262_quirks.c
@@ -10,13 +10,7 @@
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,
@@ -66,163 +60,30 @@
{ } /* end */
};
-/* update HP, line and mono-out pins according to the master switch */
-#define alc262_hp_master_update alc260_hp_master_update
-
-static void alc262_hp_bpc_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static void alc262_hp_wildwest_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- 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
+
+static int alc262_hippo_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 alc262_hippo_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;
+ update_outputs(codec);
+ return 1;
+}
#define ALC262_HIPPO_MASTER_SWITCH \
{ \
@@ -239,6 +100,9 @@
(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),
@@ -279,8 +143,7 @@
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc262_hippo1_setup(struct hda_codec *codec)
@@ -289,8 +152,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
@@ -353,8 +215,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
@@ -496,8 +357,7 @@
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN);
}
/*
@@ -571,27 +431,6 @@
},
};
-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;
@@ -599,8 +438,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.hp_pins[1] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* bind volumes of both NID 0x0c and 0x0d */
@@ -646,8 +484,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
@@ -752,8 +589,8 @@
mute = 0;
/* auto-mute only when HP is used as HP */
if (!spec->cur_mux[0]) {
- spec->jack_present = snd_hda_jack_detect(codec, 0x15);
- if (spec->jack_present)
+ spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15);
+ if (spec->hp_jack_present)
mute = HDA_AMP_MUTE;
}
/* mute/unmute internal speaker */
@@ -817,206 +654,6 @@
{ } /* 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 */
@@ -1042,13 +679,8 @@
[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",
@@ -1061,41 +693,6 @@
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),
@@ -1166,68 +763,6 @@
.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 },
@@ -1238,19 +773,6 @@
.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/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
deleted file mode 100644
index be58bf2..0000000
--- a/sound/pci/hda/alc268_quirks.c
+++ /dev/null
@@ -1,636 +0,0 @@
-/*
- * 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/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
deleted file mode 100644
index 14fdcf2..0000000
--- a/sound/pci/hda/alc269_quirks.c
+++ /dev/null
@@ -1,681 +0,0 @@
-/*
- * 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/sound/pci/hda/alc662_quirks.c b/sound/pci/hda/alc662_quirks.c
deleted file mode 100644
index e69a6ea..0000000
--- a/sound/pci/hda/alc662_quirks.c
+++ /dev/null
@@ -1,1408 +0,0 @@
-/*
- * 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/sound/pci/hda/alc680_quirks.c b/sound/pci/hda/alc680_quirks.c
deleted file mode 100644
index 0eeb227..0000000
--- a/sound/pci/hda/alc680_quirks.c
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * 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/sound/pci/hda/alc861_quirks.c b/sound/pci/hda/alc861_quirks.c
deleted file mode 100644
index d719ec6..0000000
--- a/sound/pci/hda/alc861_quirks.c
+++ /dev/null
@@ -1,725 +0,0 @@
-/*
- * 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/sound/pci/hda/alc861vd_quirks.c b/sound/pci/hda/alc861vd_quirks.c
deleted file mode 100644
index 8f28450..0000000
--- a/sound/pci/hda/alc861vd_quirks.c
+++ /dev/null
@@ -1,605 +0,0 @@
-/*
- * 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/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
index c844d2b..bea22ed 100644
--- a/sound/pci/hda/alc880_quirks.c
+++ b/sound/pci/hda/alc880_quirks.c
@@ -749,8 +749,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc880_uniwill_init_hook(struct hda_codec *codec)
@@ -781,8 +780,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1051,8 +1049,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/*
@@ -1137,8 +1134,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -1188,7 +1184,7 @@
struct alc_spec *spec = codec->spec;
alc_hp_automute(codec);
/* toggle EAPD */
- if (spec->jack_present)
+ if (spec->hp_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);
@@ -1210,8 +1206,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
index 617d047..e251514 100644
--- a/sound/pci/hda/alc882_quirks.c
+++ b/sound/pci/hda/alc882_quirks.c
@@ -173,8 +173,7 @@
spec->autocfg.speaker_pins[2] = 0x17;
spec->autocfg.speaker_pins[3] = 0x19;
spec->autocfg.speaker_pins[4] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc889_intel_init_hook(struct hda_codec *codec)
@@ -191,8 +190,7 @@
spec->autocfg.hp_pins[1] = 0x1b; /* hp */
spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
spec->autocfg.speaker_pins[1] = 0x15; /* bass */
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/*
@@ -475,8 +473,7 @@
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
@@ -487,8 +484,7 @@
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
@@ -499,8 +495,7 @@
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
@@ -511,8 +506,7 @@
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
#define ALC882_DIGOUT_NID 0x06
@@ -1711,8 +1705,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
#define alc885_mb5_setup alc885_imac24_setup
@@ -1721,12 +1714,11 @@
/* 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;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
@@ -1737,8 +1729,7 @@
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc885_imac91_setup(struct hda_codec *codec)
@@ -1748,8 +1739,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc882_targa_verbs[] = {
@@ -1773,7 +1763,7 @@
struct alc_spec *spec = codec->spec;
alc_hp_automute(codec);
snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- spec->jack_present ? 1 : 3);
+ spec->hp_jack_present ? 1 : 3);
}
static void alc882_targa_setup(struct hda_codec *codec)
@@ -1782,8 +1772,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -2187,8 +2176,7 @@
spec->autocfg.hp_pins[0] = 0x1a;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
@@ -2341,8 +2329,7 @@
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc883_mitac_verbs[] = {
@@ -2507,8 +2494,7 @@
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x18;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc888_3st_hp_verbs[] = {
@@ -2568,8 +2554,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.line_out_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* toggle speaker-output according to the hp-jack state */
@@ -2579,8 +2564,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* toggle speaker-output according to the hp-jack state */
@@ -2593,8 +2577,7 @@
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
@@ -2623,8 +2606,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc883_haier_w66_setup(struct hda_codec *codec)
@@ -2633,8 +2615,7 @@
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc883_lenovo_101e_setup(struct hda_codec *codec)
@@ -2644,10 +2625,7 @@
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;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
/* toggle speaker-output according to the hp-jack state */
@@ -2658,8 +2636,7 @@
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -2689,8 +2666,7 @@
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc888_lenovo_sky_setup(struct hda_codec *codec)
@@ -2703,8 +2679,7 @@
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
spec->autocfg.speaker_pins[4] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static void alc883_vaiott_setup(struct hda_codec *codec)
@@ -2714,8 +2689,7 @@
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc888_asus_m90v_verbs[] = {
@@ -2739,8 +2713,7 @@
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
+ alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
}
static const struct hda_verb alc888_asus_eee1601_verbs[] = {
diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c
index 2be1129..a18952e 100644
--- a/sound/pci/hda/alc_quirks.c
+++ b/sound/pci/hda/alc_quirks.c
@@ -453,6 +453,19 @@
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/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..1715e8b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -34,6 +34,9 @@
#include "hda_beep.h"
#include <sound/hda_hwdep.h>
+#define CREATE_TRACE_POINTS
+#include "hda_trace.h"
+
/*
* vendor / preset table
*/
@@ -208,15 +211,19 @@
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;
@@ -579,9 +586,13 @@
return -1;
}
recursive++;
- for (i = 0; i < nums; i++)
+ 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;
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
+ }
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
@@ -603,6 +614,7 @@
struct hda_bus_unsolicited *unsol;
unsigned int wp;
+ trace_hda_unsol_event(bus, res, res_ex);
unsol = bus->unsol;
if (!unsol)
return 0;
@@ -1479,8 +1491,11 @@
struct hda_cvt_setup *q)
{
hda_nid_t nid = q->nid;
- 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);
+ 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
+);
memset(q, 0, sizeof(*q));
q->nid = nid;
}
@@ -1685,6 +1700,29 @@
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
* @nid: the pin NID to sense
@@ -4083,6 +4121,7 @@
return;
}
+ trace_hda_power_down(codec);
hda_call_codec_suspend(codec);
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
@@ -4121,6 +4160,7 @@
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;
@@ -4533,6 +4573,11 @@
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,
@@ -4565,6 +4610,10 @@
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,
@@ -4645,6 +4694,27 @@
}
}
+/* 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
*
@@ -4662,12 +4732,13 @@
* The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
* respectively.
*/
-int snd_hda_parse_pin_def_config(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids)
+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)
{
hda_nid_t nid, end_nid;
- short seq, assoc_line_out, assoc_speaker;
+ short seq, assoc_line_out;
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)];
@@ -4678,7 +4749,7 @@
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 = assoc_speaker = 0;
+ assoc_line_out = 0;
end_nid = codec->start_nid + codec->num_nodes;
for (nid = codec->start_nid; nid < end_nid; nid++) {
@@ -4730,16 +4801,10 @@
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] = seq;
+ sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
cfg->speaker_outs++;
break;
case AC_JACK_HP_OUT:
@@ -4788,7 +4853,8 @@
* 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) {
+ if (!cfg->line_outs && cfg->hp_outs > 1 &&
+ !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
int i = 0;
while (i < cfg->hp_outs) {
/* The real HPs should have the sequence 0x0f */
@@ -4825,7 +4891,8 @@
* FIX-UP: if no line-outs are detected, try to use speaker or HP pin
* as a primary output
*/
- if (!cfg->line_outs) {
+ if (!cfg->line_outs &&
+ !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
if (cfg->speaker_outs) {
cfg->line_outs = cfg->speaker_outs;
memcpy(cfg->line_out_pins, cfg->speaker_pins,
@@ -4843,21 +4910,9 @@
}
}
- /* 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;
- }
+ 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);
sort_autocfg_input_pins(cfg);
@@ -4895,7 +4950,7 @@
return 0;
}
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config);
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
int snd_hda_get_input_pin_attr(unsigned int def_conf)
{
@@ -5154,30 +5209,6 @@
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
* @buf: the string buffer to write
@@ -5218,6 +5249,8 @@
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/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..1c8ddf5 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(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
-
- 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;
+ return val;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -326,6 +318,11 @@
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;
@@ -335,24 +332,59 @@
snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n");
size = 128;
}
- if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) {
+ if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) {
snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size);
return -ERANGE;
}
- buf = kmalloc(size, GFP_KERNEL);
- if (!buf)
- return -ENOMEM;
+ /* set ELD buffer */
+ buf = eld->eld_buffer;
- for (i = 0; i < size; i++)
- buf[i] = hdmi_get_eld_byte(codec, nid, i);
+ 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;
+ }
ret = hdmi_update_eld(eld, buf, size);
- kfree(buf);
+error:
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];
@@ -361,7 +393,7 @@
if (!a->format)
return;
- snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+ hdmi_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);
@@ -420,7 +452,7 @@
i, a->format, cea_audio_coding_type_names[a->format]);
snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels);
- snd_print_pcm_rates(a->rates, buf, sizeof(buf));
+ hdmi_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/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index bf3ced5..72e5885 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -643,14 +643,14 @@
static void parse_codec_mode(char *buf, struct hda_bus *bus,
struct hda_codec **codecp)
{
- unsigned int vendorid, subid, caddr;
+ 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 (codec->vendor_id == vendorid &&
- codec->subsystem_id == subid &&
+ if ((vendorid <= 0 || codec->vendor_id == vendorid) &&
+ (subid <= 0 || codec->subsystem_id == subid) &&
codec->addr == caddr) {
*codecp = codec;
break;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index be69822..bd7fc99 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -34,7 +34,6 @@
*
*/
-#include <asm/io.h>
#include <linux/delay.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
@@ -46,6 +45,12 @@
#include <linux/pci.h>
#include <linux/mutex.h>
#include <linux/reboot.h>
+#include <linux/io.h>
+#ifdef CONFIG_X86
+/* for snoop control */
+#include <asm/pgtable.h>
+#include <asm/cacheflush.h>
+#endif
#include <sound/core.h>
#include <sound/initval.h>
#include "hda_codec.h"
@@ -116,6 +121,22 @@
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},"
@@ -360,7 +381,7 @@
*/
unsigned char stream_tag; /* assigned stream */
unsigned char index; /* stream index */
- int device; /* last device number assigned to */
+ int assigned_key; /* last device# key assigned to */
unsigned int opened :1;
unsigned int running :1;
@@ -371,6 +392,7 @@
* when link position is not greater than FIFO size
*/
unsigned int insufficient :1;
+ unsigned int wc_marked:1;
};
/* CORB/RIRB */
@@ -438,6 +460,7 @@
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];
@@ -481,6 +504,7 @@
#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 \
@@ -542,6 +566,45 @@
/* 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);
/*
@@ -563,6 +626,7 @@
snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
return err;
}
+ mark_pages_wc(chip, &chip->rb, true);
return 0;
}
@@ -1079,7 +1143,15 @@
static void azx_init_pci(struct azx *chip)
{
- unsigned short snoop;
+ /* 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");
+ }
+ }
/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
* TCSEL == Traffic Class Select Register, which sets PCI express QOS
@@ -1096,15 +1168,15 @@
* we need to enable snoop.
*/
if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) {
- snd_printdd(SFX "Enabling ATI snoop\n");
+ snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip));
update_pci_byte(chip->pci,
- ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07,
+ azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0);
}
/* For NVIDIA HDA, enable snoop */
if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) {
- snd_printdd(SFX "Enabling Nvidia snoop\n");
+ snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip));
update_pci_byte(chip->pci,
NVIDIA_HDA_TRANSREG_ADDR,
0x0f, NVIDIA_HDA_ENABLE_COHBITS);
@@ -1118,16 +1190,20 @@
/* 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 (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
- pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC,
- snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
+ 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);
pci_read_config_word(chip->pci,
INTEL_SCH_HDA_DEVC, &snoop);
- snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n",
- (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
- ? "Failed" : "OK");
}
+ snd_printdd(SFX "SCH snoop: %s\n",
+ (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
+ ? "Disabled" : "Enabled");
}
}
@@ -1334,12 +1410,16 @@
*/
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 */
- 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));
+ 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);
/* program the length of samples in cyclic buffer */
azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize);
@@ -1533,6 +1613,9 @@
{
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;
@@ -1544,12 +1627,12 @@
for (i = 0; i < nums; i++, dev++)
if (!chip->azx_dev[dev].opened) {
res = &chip->azx_dev[dev];
- if (res->device == substream->pcm->device)
+ if (res->assigned_key == key)
break;
}
if (res) {
res->opened = 1;
- res->device = substream->pcm->device;
+ res->assigned_key = key;
}
return res;
}
@@ -1599,6 +1682,7 @@
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);
@@ -1613,10 +1697,25 @@
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,
- 128);
+ buff_step);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
- 128);
+ buff_step);
snd_hda_power_up(apcm->codec);
err = hinfo->ops.open(hinfo, apcm->codec, substream);
if (err < 0) {
@@ -1671,19 +1770,30 @@
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;
- return snd_pcm_lib_malloc_pages(substream,
+ ret = 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 */
@@ -1696,6 +1806,7 @@
snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
+ mark_runtime_wc(chip, azx_dev, runtime, false);
return snd_pcm_lib_free_pages(substream);
}
@@ -1924,7 +2035,8 @@
}
static unsigned int azx_get_position(struct azx *chip,
- struct azx_dev *azx_dev)
+ struct azx_dev *azx_dev,
+ bool with_check)
{
unsigned int pos;
int stream = azx_dev->substream->stream;
@@ -1940,7 +2052,7 @@
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
- if (chip->position_fix[stream] == POS_FIX_AUTO) {
+ if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos || pos == (u32)-1) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
@@ -1964,7 +2076,7 @@
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));
+ azx_get_position(chip, azx_dev, false));
}
/*
@@ -1987,7 +2099,7 @@
return -1; /* bogus (too early) interrupt */
stream = azx_dev->substream->stream;
- pos = azx_get_position(chip, azx_dev);
+ pos = azx_get_position(chip, azx_dev, true);
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@@ -2054,6 +2166,20 @@
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,
@@ -2063,6 +2189,7 @@
.prepare = azx_pcm_prepare,
.trigger = azx_pcm_trigger,
.pointer = azx_pcm_pointer,
+ .mmap = azx_pcm_mmap,
.page = snd_pcm_sgbuf_ops_page,
};
@@ -2343,13 +2470,19 @@
if (chip->azx_dev) {
for (i = 0; i < chip->num_streams; i++)
- if (chip->azx_dev[i].bdl.area)
+ if (chip->azx_dev[i].bdl.area) {
+ mark_pages_wc(chip, &chip->azx_dev[i].bdl, false);
snd_dma_free_pages(&chip->azx_dev[i].bdl);
+ }
}
- if (chip->rb.area)
+ if (chip->rb.area) {
+ mark_pages_wc(chip, &chip->rb, false);
snd_dma_free_pages(&chip->rb);
- if (chip->posbuf.area)
+ }
+ if (chip->posbuf.area) {
+ mark_pages_wc(chip, &chip->posbuf, false);
snd_dma_free_pages(&chip->posbuf);
+ }
pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip->azx_dev);
@@ -2369,6 +2502,7 @@
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),
@@ -2544,6 +2678,7 @@
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
+ chip->snoop = hda_snoop;
if (bdl_pos_adj[dev] < 0) {
switch (chip->driver_type) {
@@ -2616,6 +2751,10 @@
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));
@@ -2667,6 +2806,7 @@
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,
@@ -2676,6 +2816,7 @@
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)
@@ -2817,37 +2958,49 @@
static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* CPT */
{ PCI_DEVICE(0x8086, 0x1c20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE },
/* PBG */
{ PCI_DEVICE(0x8086, 0x1d20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE},
/* Panther Point */
{ PCI_DEVICE(0x8086, 0x1e20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE},
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
- .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
+ .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE},
{ PCI_DEVICE(0x8086, 0x2668),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH6 */
{ PCI_DEVICE(0x8086, 0x27d8),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH7 */
{ PCI_DEVICE(0x8086, 0x269a),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ESB2 */
{ PCI_DEVICE(0x8086, 0x284b),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH8 */
{ PCI_DEVICE(0x8086, 0x293e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x293f),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x3a3e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* ICH10 */
{ PCI_DEVICE(0x8086, 0x3a6e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
+ AZX_DCAPS_BUFSIZE }, /* 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 },
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE },
/* ATI SB 450/600/700/800/900 */
{ PCI_DEVICE(0x1002, 0x437b),
.driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 2e7ac31..46c581c 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -267,11 +267,14 @@
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 extra_out_nid[3]; /* optional DACs, 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 dig_out_nid; /* digital out audio widget */
const hda_nid_t *slave_dig_outs;
int max_channels; /* currently supported analog channels */
@@ -333,9 +336,6 @@
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 @@
AUTO_PIN_HP_OUT
};
-#define AUTO_CFG_MAX_OUTS 5
+#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
#define AUTO_CFG_MAX_INS 8
struct auto_pin_cfg_item {
@@ -443,9 +443,18 @@
#define get_defcfg_device(cfg) \
((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
-int snd_hda_parse_pin_def_config(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids);
+/* 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)
/* amp values */
#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
@@ -492,6 +501,8 @@
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);
@@ -607,6 +618,7 @@
};
#define ELD_FIXED_BYTES 20
+#define ELD_MAX_SIZE 256
#define ELD_MAX_MNL 16
#define ELD_MAX_SAD 16
@@ -631,6 +643,7 @@
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/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 2be57b0..2c981b5 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -152,12 +152,18 @@
static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
{
- char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
+ static unsigned int rates[] = {
+ 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000, 384000
+ };
+ int i;
pcm &= AC_SUPPCM_RATES;
snd_iprintf(buffer, " rates [0x%x]:", pcm);
- snd_print_pcm_rates(pcm, buf, sizeof(buf));
- snd_iprintf(buffer, "%s\n", buf);
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
+ if (pcm & (1 << i))
+ snd_iprintf(buffer, " %d", rates[i]);
+ snd_iprintf(buffer, "\n");
}
static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h
new file mode 100644
index 0000000..9884871
--- /dev/null
+++ b/sound/pci/hda/hda_trace.h
@@ -0,0 +1,117 @@
+#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 <linux/tracepoint.h>
+
+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 <trace/define_trace.h>
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 8648917..d8aac58 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -48,6 +48,8 @@
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;
@@ -302,6 +304,72 @@
}
#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
*/
@@ -310,8 +378,15 @@
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ int err;
+ set_stream_active(codec, true);
+ err = 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,
@@ -333,11 +408,41 @@
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,
- /* NID is set in ad198x_build_pcms */
+ .ops = {
+ .open = ad1988_alt_playback_pcm_open,
+ .close = ad1988_alt_playback_pcm_close
+ },
};
/*
@@ -402,7 +507,6 @@
return 0;
}
-
/*
*/
static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
@@ -413,7 +517,8 @@
.ops = {
.open = ad198x_playback_pcm_open,
.prepare = ad198x_playback_pcm_prepare,
- .cleanup = ad198x_playback_pcm_cleanup
+ .cleanup = ad198x_playback_pcm_cleanup,
+ .close = ad198x_playback_pcm_close
},
};
@@ -2058,7 +2163,6 @@
enum {
AD1988_6STACK,
AD1988_6STACK_DIG,
- AD1988_6STACK_DIG_FP,
AD1988_3STACK,
AD1988_3STACK_DIG,
AD1988_LAPTOP,
@@ -2168,6 +2272,17 @@
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),
@@ -2188,6 +2303,7 @@
};
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),
@@ -2210,13 +2326,6 @@
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 */
};
@@ -2238,6 +2347,7 @@
};
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),
@@ -2272,6 +2382,7 @@
/* 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),
@@ -2446,7 +2557,7 @@
{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, 0x01}, /* DAC1:04h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
{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},
@@ -2594,7 +2705,7 @@
{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, 0x01}, /* DAC1:04h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
{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},
@@ -2669,7 +2780,7 @@
{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, 0x01}, /* DAC1:04h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
{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},
@@ -2782,11 +2893,11 @@
{
static const hda_nid_t idx_to_dac[8] = {
/* A B C D E F G H */
- 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
+ 0x03, 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 */
- 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
+ 0x03, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
};
if (is_rev2(codec))
return idx_to_dac_rev2[idx];
@@ -3023,8 +3134,8 @@
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 04 */
- snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
+ case 0x11: /* port-A - DAC 03 */
+ snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x00);
break;
case 0x14: /* port-B - DAC 06 */
snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02);
@@ -3150,7 +3261,6 @@
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",
@@ -3208,10 +3318,11 @@
}
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))
@@ -3227,19 +3338,7 @@
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_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)) {
+ if (board_config == AD1988_6STACK_DIG) {
spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
spec->dig_in_nid = AD1988_SPDIF_IN;
}
@@ -3282,6 +3381,15 @@
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/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i;
+ int i, idx;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,9 +384,11 @@
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
- if (*idxp >= 0)
+ idx = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (idx >= 0) {
+ *idxp = idx;
return nid;
+ }
}
return 0;
}
@@ -533,7 +535,7 @@
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[32];
+ char tmp[44];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..686ec6d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1867,39 +1867,6 @@
{ } /* 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},
@@ -1968,7 +1935,6 @@
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 */
@@ -1980,7 +1946,6 @@
[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",
@@ -1995,7 +1960,6 @@
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),
{}
};
@@ -2053,13 +2017,6 @@
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;
@@ -3110,6 +3067,7 @@
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),
@@ -3348,6 +3306,8 @@
#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)
{
@@ -3370,16 +3330,26 @@
/* 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 type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ 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]);
}
return nums;
}
@@ -3395,19 +3365,19 @@
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,
- 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, 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);
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)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3832,7 @@
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4005,8 @@
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) {
@@ -4167,9 +4139,11 @@
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ 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, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4165,7 @@
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4184,7 @@
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
@@ -4378,6 +4351,53 @@
.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;
@@ -4391,6 +4411,9 @@
return -ENOMEM;
codec->spec = spec;
codec->pin_amp_workaround = 1;
+
+ apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
+
err = cx_auto_search_adcs(codec);
if (err < 0)
return err;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 19cb72d..3425401 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -324,6 +324,66 @@
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)
@@ -967,19 +1027,12 @@
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;
}
@@ -1162,6 +1215,25 @@
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;
@@ -1170,12 +1242,25 @@
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;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60..011644b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -159,23 +159,27 @@
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 jack_present: 1;
+ unsigned int hp_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: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 */
+ 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 */
/* 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;
@@ -193,6 +197,7 @@
/* 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;
@@ -202,6 +207,9 @@
/* 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 */
@@ -525,8 +533,8 @@
}
}
-/* Toggle internal speakers muting */
-static void update_speakers(struct hda_codec *codec)
+/* Toggle outputs muting */
+static void update_outputs(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int on;
@@ -538,10 +546,10 @@
do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins, spec->master_mute, true);
- if (!spec->automute)
+ if (!spec->automute_speaker)
on = 0;
else
- on = spec->jack_present | spec->line_jack_present;
+ on = spec->hp_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);
@@ -551,26 +559,35 @@
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_lines || !spec->automute)
+ if (!spec->automute_lo)
on = 0;
else
- on = spec->jack_present;
+ on = spec->hp_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;
- if (!spec->automute)
- return;
- spec->jack_present =
+ spec->hp_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
- update_speakers(codec);
+ if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo))
+ return;
+ call_update_outputs(codec);
}
/* standard line-out-automute helper */
@@ -578,12 +595,16 @@
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
+ /* check LO jack only when it's different from HP */
+ if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
return;
+
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
- update_speakers(codec);
+ if (!spec->automute_speaker || !spec->detect_lo)
+ return;
+ call_update_outputs(codec);
}
#define get_connection_index(codec, mux, nid) \
@@ -781,7 +802,7 @@
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- if (spec->automute_hp_lo) {
+ if (spec->automute_speaker_possible && spec->automute_lo_possible) {
uinfo->value.enumerated.items = 3;
texts = texts3;
} else {
@@ -800,13 +821,12 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- unsigned int val;
- if (!spec->automute)
- val = 0;
- else if (!spec->automute_lines)
- val = 1;
- else
- val = 2;
+ unsigned int val = 0;
+ if (spec->automute_speaker)
+ val++;
+ if (spec->automute_lo)
+ val++;
+
ucontrol->value.enumerated.item[0] = val;
return 0;
}
@@ -819,28 +839,36 @@
switch (ucontrol->value.enumerated.item[0]) {
case 0:
- if (!spec->automute)
+ if (!spec->automute_speaker && !spec->automute_lo)
return 0;
- spec->automute = 0;
+ spec->automute_speaker = 0;
+ spec->automute_lo = 0;
break;
case 1:
- if (spec->automute && !spec->automute_lines)
- return 0;
- spec->automute = 1;
- spec->automute_lines = 0;
+ 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;
break;
case 2:
- if (!spec->automute_hp_lo)
+ if (!spec->automute_lo_possible || !spec->automute_speaker_possible)
return -EINVAL;
- if (spec->automute && spec->automute_lines)
+ if (spec->automute_speaker && spec->automute_lo)
return 0;
- spec->automute = 1;
- spec->automute_lines = 1;
+ spec->automute_speaker = 1;
+ spec->automute_lo = 1;
break;
default:
return -EINVAL;
}
- update_speakers(codec);
+ call_update_outputs(codec);
return 1;
}
@@ -877,7 +905,7 @@
* Check the availability of HP/line-out auto-mute;
* Set up appropriately if really supported
*/
-static void alc_init_auto_hp(struct hda_codec *codec)
+static void alc_init_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
@@ -892,8 +920,6 @@
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) {
@@ -909,6 +935,8 @@
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))
@@ -918,28 +946,32 @@
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC_HP_EVENT);
- 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_lines = spec->detect_line;
+ spec->detect_hp = 1;
}
- if (spec->automute) {
+ 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_lo_possible = spec->detect_hp;
+ }
+
+ 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) {
/* create a control for automute mode */
alc_add_automute_mode_enum(codec);
spec->unsol_event = alc_sku_unsol_event;
@@ -1140,7 +1172,7 @@
/* check the availabilities of auto-mute and auto-mic switches */
static void alc_auto_check_switches(struct hda_codec *codec)
{
- alc_init_auto_hp(codec);
+ alc_init_automute(codec);
alc_init_auto_mic(codec);
}
@@ -1320,7 +1352,9 @@
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!spec->autocfg.hp_pins[0]) {
+ if (!spec->autocfg.hp_pins[0] &&
+ !(spec->autocfg.line_out_pins[0] &&
+ spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
@@ -1521,6 +1555,15 @@
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
*/
@@ -1784,6 +1827,7 @@
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
+ "PCM Playback Volume",
NULL,
};
@@ -1798,6 +1842,7 @@
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
+ "PCM Playback Switch",
NULL,
};
@@ -2359,6 +2404,18 @@
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;
@@ -2369,6 +2426,7 @@
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);
}
@@ -2432,6 +2490,47 @@
}
/*
+ * 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
*/
@@ -2439,11 +2538,15 @@
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 */
@@ -2484,13 +2587,14 @@
#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 &&
@@ -2513,7 +2617,10 @@
return "PCM";
break;
}
- return chname[ch];
+ if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name)))
+ return "PCM";
+
+ return channel_name[ch];
}
/* create input playback/capture controls for the given pin */
@@ -2777,8 +2884,9 @@
if (found_in_nid_list(nid, spec->multiout.dac_nids,
spec->multiout.num_dacs))
continue;
- if (spec->multiout.hp_nid == nid)
- continue;
+ if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
+ ARRAY_SIZE(spec->multiout.hp_out_nid)))
+ continue;
if (found_in_nid_list(nid, spec->multiout.extra_out_nid,
ARRAY_SIZE(spec->multiout.extra_out_nid)))
continue;
@@ -2795,6 +2903,29 @@
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)
{
@@ -2806,7 +2937,7 @@
again:
/* set num_dacs once to full for alc_auto_look_for_dac() */
spec->multiout.num_dacs = cfg->line_outs;
- spec->multiout.hp_nid = 0;
+ spec->multiout.hp_out_nid[0] = 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;
@@ -2817,7 +2948,7 @@
spec->private_dac_nids[i] =
get_dac_if_single(codec, cfg->line_out_pins[i]);
if (cfg->hp_outs)
- spec->multiout.hp_nid =
+ spec->multiout.hp_out_nid[0] =
get_dac_if_single(codec, cfg->hp_pins[0]);
if (cfg->speaker_outs)
spec->multiout.extra_out_nid[0] =
@@ -2849,12 +2980,28 @@
sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
}
- 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]);
+ 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);
return 0;
}
@@ -2955,7 +3102,7 @@
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) {
+ if (!name || !strcmp(name, "CLFE")) {
/* Center/LFE */
err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1);
if (err < 0)
@@ -2981,16 +3128,13 @@
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) {
/* the corresponding DAC is already occupied */
if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
@@ -3011,20 +3155,108 @@
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)
+ return alc_auto_create_extra_out(codec, *pins, *dacs, 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_out(codec, spec->autocfg.hp_pins[0],
- spec->multiout.hp_nid,
- "Headphone");
+ return alc_auto_create_extra_outs(codec, spec->autocfg.hp_outs,
+ spec->autocfg.hp_pins,
+ spec->multiout.hp_out_nid,
+ "Headphone");
}
static int alc_auto_create_speaker_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- return alc_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0],
- spec->multiout.extra_out_nid[0],
- "Speaker");
+ return alc_auto_create_extra_outs(codec, spec->autocfg.speaker_outs,
+ spec->autocfg.speaker_pins,
+ spec->multiout.extra_out_nid,
+ "Speaker");
}
static void alc_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -3081,16 +3313,39 @@
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ int i;
+ hda_nid_t pin, 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]);
+ 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);
+ }
}
/*
@@ -3101,6 +3356,7 @@
{
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--) {
@@ -3128,8 +3384,13 @@
}
}
spec->multiout.num_dacs = 1;
- if (num_pins < 2)
+ 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;
return 0;
+ }
return num_pins;
}
@@ -3215,36 +3476,11 @@
.put = alc_auto_ch_mode_put,
};
-static int alc_auto_add_multi_channel_mode(struct hda_codec *codec,
- int (*fill_dac)(struct hda_codec *))
+static int alc_auto_add_multi_channel_mode(struct hda_codec *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;
-
- 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) {
+ if (spec->multi_ios > 0) {
struct snd_kcontrol_new *knew;
knew = alc_kcontrol_new(spec);
@@ -3254,10 +3490,6 @@
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;
}
@@ -3540,27 +3772,42 @@
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_def_config(codec, &spec->autocfg,
- ignore_nids);
+ err = snd_hda_parse_pin_defcfg(codec, cfg, ignore_nids,
+ spec->parse_flags);
if (err < 0)
return err;
- if (!spec->autocfg.line_outs) {
- if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+ if (!cfg->line_outs) {
+ if (cfg->dig_outs || cfg->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, alc_auto_fill_dac_nids);
+ err = alc_auto_add_multi_channel_mode(codec);
if (err < 0)
return err;
- err = alc_auto_create_multi_out_ctls(codec, &spec->autocfg);
+ err = alc_auto_create_multi_out_ctls(codec, cfg);
if (err < 0)
return err;
err = alc_auto_create_hp_out(codec);
@@ -3663,10 +3910,8 @@
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc880_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -3691,10 +3936,8 @@
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -3709,6 +3952,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
@@ -3790,10 +4037,8 @@
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc260_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -3818,10 +4063,8 @@
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
}
@@ -3839,6 +4082,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
@@ -3865,6 +4112,7 @@
PINFIX_LENOVO_Y530,
PINFIX_PB_M5210,
PINFIX_ACER_ASPIRE_7736,
+ PINFIX_ASUS_W90V,
};
static const struct alc_fixup alc882_fixups[] = {
@@ -3896,10 +4144,18 @@
.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),
@@ -3946,6 +4202,10 @@
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);
@@ -3969,10 +4229,8 @@
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc882_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -3997,10 +4255,8 @@
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -4019,6 +4275,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
@@ -4123,10 +4383,8 @@
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -4151,10 +4409,8 @@
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -4174,6 +4430,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
/*
@@ -4222,14 +4482,9 @@
/*
*/
-#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);
@@ -4240,38 +4495,10 @@
/* ALC268 has no aa-loopback mixer */
- 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]);
+ /* automatic parse from the BIOS config */
+ err = alc268_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
has_beep = 0;
for (i = 0; i < spec->num_mixers; i++) {
@@ -4283,10 +4510,8 @@
if (has_beep) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
/* override the amp caps for beep generator */
snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
@@ -4308,13 +4533,16 @@
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
alc_init_jacks(codec);
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
/*
@@ -4408,9 +4636,9 @@
static void alc269_shutup(struct hda_codec *codec)
{
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
alc269_toggle_power_output(codec, 0);
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
@@ -4419,19 +4647,19 @@
#ifdef CONFIG_PM
static int alc269_resume(struct hda_codec *codec)
{
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
codec->patch_ops.init(codec);
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
alc269_toggle_power_output(codec, 1);
msleep(200);
}
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018)
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x018)
alc269_toggle_power_output(codec, 1);
snd_hda_codec_resume_amp(codec);
@@ -4484,6 +4712,46 @@
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,
@@ -4494,6 +4762,13 @@
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[] = {
@@ -4556,23 +4831,144 @@
.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"},
{}
};
@@ -4581,23 +4977,23 @@
{
int val;
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) {
+ if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8817);
}
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x016) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8814);
}
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
val = alc_read_coef_idx(codec, 0x04);
/* Power up output pin */
alc_write_coef_idx(codec, 0x04, val | (1<<11));
}
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
val = alc_read_coef_idx(codec, 0xd);
if ((val & 0x0c00) >> 10 != 0x1) {
/* Capless ramp up clock control */
@@ -4621,15 +5017,10 @@
/*
*/
-#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 board_config, coef;
- int err;
+ int err = 0;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4641,72 +5032,41 @@
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;
- coef = alc_read_coef_idx(codec, 0);
- if ((coef & 0x00f0) == 0x0010) {
+ switch (alc_get_coef0(codec) & 0x00f0) {
+ case 0x0010:
if (codec->bus->pci->subsystem_vendor == 0x1025 &&
- 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->cdefine.platform_type == 1)
+ err = alc_codec_rename(codec, "ALC271X");
spec->codec_variant = ALC269_TYPE_ALC269VB;
- } 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");
+ break;
+ case 0x0020:
+ if (codec->bus->pci->subsystem_vendor == 0x17aa &&
+ codec->bus->pci->subsystem_device == 0x21f3)
+ err = alc_codec_rename(codec, "ALC3202");
spec->codec_variant = ALC269_TYPE_ALC269VC;
- } else
+ break;
+ default:
alc_fix_pll_init(codec, 0x20, 0x04, 15);
+ }
+ if (err < 0)
+ goto error;
alc269_fill_coef(codec);
}
- board_config = alc_board_config(codec, ALC269_MODEL_LAST,
- alc269_models, alc269_cfg_tbl);
+ alc_pick_fixup(codec, alc269_fixup_models,
+ alc269_fixup_tbl, alc269_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- 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]);
+ /* automatic parse from the BIOS config */
+ err = alc269_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -4719,10 +5079,8 @@
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
}
@@ -4734,8 +5092,7 @@
#ifdef CONFIG_PM
codec->patch_ops.resume = alc269_resume;
#endif
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc269_shutup;
alc_init_jacks(codec);
@@ -4747,6 +5104,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
/*
@@ -4794,14 +5155,9 @@
/*
*/
-#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);
@@ -4812,39 +5168,13 @@
spec->mixer_nid = 0x15;
- board_config = alc_board_config(codec, ALC861_MODEL_LAST,
- alc861_models, alc861_cfg_tbl);
+ alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- 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]);
+ /* automatic parse from the BIOS config */
+ err = alc861_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -4857,10 +5187,8 @@
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
}
@@ -4869,18 +5197,18 @@
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO) {
- spec->init_hook = alc_auto_init_std;
+ 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;
}
/*
@@ -4902,24 +5230,41 @@
}
enum {
- ALC660VD_FIX_ASUS_GPIO1
+ ALC660VD_FIX_ASUS_GPIO1,
+ ALC861VD_FIX_DALLAS,
};
-/* reset 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);
+ }
+}
+
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),
{}
};
@@ -4931,14 +5276,10 @@
/*
*/
-#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, board_config;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4948,39 +5289,13 @@
spec->mixer_nid = 0x0b;
- board_config = alc_board_config(codec, ALC861VD_MODEL_LAST,
- alc861vd_models, alc861vd_cfg_tbl);
+ alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- 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]);
+ /* automatic parse from the BIOS config */
+ err = alc861vd_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
@@ -4998,10 +5313,8 @@
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -5011,8 +5324,7 @@
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -5020,6 +5332,10 @@
#endif
return 0;
+
+ error:
+ alc_free(codec);
+ return err;
}
/*
@@ -5077,6 +5393,14 @@
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[] = {
@@ -5118,37 +5442,204 @@
.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, board_config;
- int coef;
+ int err = 0;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (!spec)
@@ -5158,50 +5649,31 @@
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);
- 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");
+ err = alc_codec_rename_from_preset(codec);
+ if (err < 0)
+ goto error;
- 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
- }
-
- if (board_config != ALC_MODEL_AUTO)
- setup_preset(codec, &alc662_presets[board_config]);
+ 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 (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -5214,10 +5686,8 @@
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
+ if (err < 0)
+ goto error;
switch (codec->vendor_id) {
case 0x10ec0662:
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -5237,8 +5707,7 @@
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
alc_init_jacks(codec);
@@ -5249,32 +5718,10 @@
#endif
return 0;
-}
-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);
+ error:
+ alc_free(codec);
+ return err;
}
/*
@@ -5288,14 +5735,9 @@
/*
*/
-#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);
@@ -5306,43 +5748,11 @@
/* ALC680 has no aa-loopback mixer */
- 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);
+ /* automatic parse from the BIOS config */
+ err = alc680_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
}
if (!spec->no_analog && !spec->cap_mixer)
@@ -5351,8 +5761,7 @@
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
+ spec->init_hook = alc_auto_init_std;
return 0;
}
@@ -5380,6 +5789,8 @@
.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 },
@@ -5392,13 +5803,13 @@
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
.patch = patch_alc882 },
- { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
+ { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
- { .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 },
+ { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b5..59a52a4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@
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)
{
@@ -696,6 +697,7 @@
return 1;
}
+#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -2970,8 +2972,9 @@
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];
+ hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac;
unsigned int wcaps, wtype;
conn_len = snd_hda_get_connections(codec, nid, conn,
@@ -2999,10 +3002,21 @@
return conn[j];
}
}
- /* if all DACs are already assigned, connect to the primary DAC */
+
+ /* 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 (conn_len > 1) {
for (j = 0; j < conn_len; j++) {
- if (conn[j] == spec->multiout.dac_nids[0]) {
+ if (conn[j] == fallback_dac) {
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, j);
break;
@@ -4128,22 +4142,14 @@
#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;
- 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 snd_hda_input_jack_add(codec, nid, type, NULL);
+#else
return 0;
+#endif /* CONFIG_SND_HDA_INPUT_JACK */
}
static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
@@ -5583,9 +5589,7 @@
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)
@@ -5628,6 +5632,7 @@
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76df:
case 0x111d76e5:
case 0x111d7666:
case 0x111d7667:
@@ -5686,22 +5691,6 @@
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;
@@ -6571,6 +6560,7 @@
{ .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},
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798..417d62a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1506,39 +1506,49 @@
struct via_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
- codec->num_pcms = 1;
+ codec->num_pcms = 0;
codec->pcm_info = info;
- 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 || 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;
- 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->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_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) {
+ 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->multiout.dig_out_nid || spec->dig_in_nid) {
snprintf(spec->stream_name_digital,
sizeof(spec->stream_name_digital),
"%s Digital", codec->chip_name);
@@ -1562,17 +1572,19 @@
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;
}
@@ -2084,7 +2096,7 @@
struct via_spec *spec = codec->spec;
struct nid_path *path;
bool check_dac;
- hda_nid_t pin, dac;
+ hda_nid_t pin, dac = 0;
int err;
pin = spec->autocfg.speaker_pins[0];
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 0ccc0eb..8531b98 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2748,8 +2748,9 @@
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),
- ice->irq, 0, &ice->rmidi[0]);
+ c->mpu401_1_info_flags |
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+ -1, &ice->rmidi[0]);
if (err < 0) {
snd_card_free(card);
return err;
@@ -2764,8 +2765,9 @@
/* 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),
- ice->irq, 0, &ice->rmidi[1]);
+ c->mpu401_2_info_flags |
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+ -1, &ice->rmidi[1]);
if (err < 0) {
snd_card_free(card);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 0378126..2fd4bf2 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2820,8 +2820,8 @@
/* 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,
- chip->irq, 0, &chip->rmidi);
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
+ -1, &chip->rmidi);
if (err < 0)
printk(KERN_WARNING "maestro3: no MIDI support.\n");
#endif
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 82311fc..53e5508 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -678,15 +678,15 @@
goto err_card;
if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) {
- unsigned int info_flags = MPU401_INFO_INTEGRATED;
+ unsigned int info_flags =
+ MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK;
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, 0, 0,
- &chip->midi);
+ info_flags, -1, &chip->midi);
if (err < 0)
goto err_card;
}
diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c
index 32d096c..8433aa7 100644
--- a/sound/pci/oxygen/xonar_pcm179x.c
+++ b/sound/pci/oxygen/xonar_pcm179x.c
@@ -1074,6 +1074,7 @@
.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/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index e34ae14..88cc776 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2109,7 +2109,7 @@
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, 0, chip->irq, 0,
+ val, MPU401_INFO_IRQ_HOOK, -1,
&chip->rmidi);
if (err < 0)
snd_printk(KERN_WARNING
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67c..6e2f7ef 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1241,10 +1241,30 @@
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 = 1 << ((hdspm_decode_latency(hdspm->control_register) + 8));
+ hdspm->period_bytes = 4 * hdspm_get_latency(hdspm);
}
@@ -1303,12 +1323,27 @@
spin_lock_irq(&s->lock);
- frames >>= 7;
- n = 0;
- while (frames) {
- n++;
- frames >>= 1;
+ 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;
+ }
}
+
s->control_register &= ~HDSPM_LatencyMask;
s->control_register |= hdspm_encode_latency(n);
@@ -1339,6 +1374,10 @@
break;
case MADIface:
freq_const = 131072000000000ULL;
+ break;
+ default:
+ snd_BUG();
+ return 0;
}
return div_u64(freq_const, period);
@@ -1356,16 +1395,19 @@
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;
+ n = 104857600000000ULL; /* 100 MHz */
+ break;
+ default:
+ snd_BUG();
+ return;
}
n = div_u64(n, rate);
@@ -4794,8 +4836,7 @@
snd_iprintf(buffer, "--- Settings ---\n");
- x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
- HDSPM_LatencyMask));
+ x = hdspm_get_latency(hdspm);
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -4958,8 +4999,7 @@
snd_iprintf(buffer, "--- Settings ---\n");
- x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
- HDSPM_LatencyMask));
+ x = hdspm_get_latency(hdspm);
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -5665,19 +5705,6 @@
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 |
@@ -5696,8 +5723,8 @@
.channels_max = HDSPM_MAX_CHANNELS,
.buffer_bytes_max =
HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
- .period_bytes_min = (64 * 4),
- .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (32 * 4),
+ .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
.periods_min = 2,
.periods_max = 512,
.fifo_size = 0
@@ -5721,31 +5748,13 @@
.channels_max = HDSPM_MAX_CHANNELS,
.buffer_bytes_max =
HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
- .period_bytes_min = (64 * 4),
- .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (32 * 4),
+ .period_bytes_max = (8192 * 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)
{
@@ -5946,26 +5955,29 @@
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_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);
-
+ 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);
break;
default:
- snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- &hw_constraints_period_sizes_old);
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ 64, 8192);
+ break;
}
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 {
@@ -6018,24 +6030,28 @@
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_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;
+ 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;
default:
- snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- &hw_constraints_period_sizes_old);
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ 64, 8192);
+ break;
}
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 {
@@ -6081,7 +6097,7 @@
}
static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
- unsigned int cmd, unsigned long __user arg)
+ unsigned int cmd, unsigned long arg)
{
void __user *argp = (void __user *)arg;
struct hdspm *hdspm = hw->private_data;
@@ -6206,11 +6222,13 @@
info.line_out = hdspm_line_out(hdspm);
info.passthru = 0;
spin_unlock_irq(&hdspm->lock);
- if (copy_to_user((void __user *) arg, &info, sizeof(info)))
+ if (copy_to_user(argp, &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);
@@ -6243,13 +6261,15 @@
break;
}
- if (copy_to_user((void __user *) arg, &status, sizeof(status)))
+ if (copy_to_user(argp, &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));
@@ -6260,13 +6280,13 @@
if (hdspm->tco)
hdspm_version.addons |= HDSPM_ADDON_TCO;
- if (copy_to_user((void __user *) arg, &hdspm_version,
+ if (copy_to_user(argp, &hdspm_version,
sizeof(hdspm_version)))
return -EFAULT;
break;
case SNDRV_HDSPM_IOCTL_GET_MIXER:
- if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
+ if (copy_from_user(&mixer, argp, sizeof(mixer)))
return -EFAULT;
if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer,
sizeof(struct hdspm_mixer)))
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index bcf6152..5ffb20b 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1234,7 +1234,7 @@
goto error;
}
- if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
KBUILD_MODNAME, sis)) {
printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
goto error;
@@ -1340,7 +1340,7 @@
if (rc)
goto error_out_cleanup;
- if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
KBUILD_MODNAME, sis)) {
printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
goto error_out_cleanup;
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 2571a67..c500816 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1493,9 +1493,10 @@
return err;
}
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES,
- sonic->midi_port, MPU401_INFO_INTEGRATED,
- sonic->irq, 0,
- &midi_uart)) < 0) {
+ sonic->midi_port,
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &midi_uart)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index d8a128f..5e707ef 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -148,8 +148,9 @@
if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
(err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
trident->midi_port,
- MPU401_INFO_INTEGRATED,
- trident->irq, 0, &trident->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &trident->rmidi)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index f03fd62..c3656ff 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1175,6 +1175,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
int err;
struct via_rate_lock *ratep;
+ bool use_src = false;
runtime->hw = snd_via82xx_hw;
@@ -1196,6 +1197,7 @@
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];
@@ -1212,6 +1214,12 @@
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;
@@ -2068,8 +2076,9 @@
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,
- chip->irq, 0, &chip->rmidi) < 0) {
+ mpu_port, MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK, -1,
+ &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/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 511d576..3253b04 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -305,8 +305,9 @@
if (chip->mpu_res) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI,
mpu_port[dev],
- MPU401_INFO_INTEGRATED,
- pci->irq, 0, &chip->rawmidi)) < 0) {
+ MPU401_INFO_INTEGRATED |
+ MPU401_INFO_IRQ_HOOK,
+ -1, &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/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index f3260e6..66ea71b 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -897,6 +897,18 @@
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)
@@ -904,11 +916,8 @@
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;
}
@@ -1013,6 +1022,18 @@
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)
@@ -1022,9 +1043,6 @@
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);
@@ -1615,7 +1633,7 @@
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 Volume", 0, YDSXGR_LEGACYOUTVOL),
+YMFPCI_DOUBLE("FM Legacy Playback 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/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 8f064c7..4080bec 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -82,7 +82,6 @@
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/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index bc823a5..775bd95 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -845,7 +845,7 @@
return ret;
}
- ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
+ ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0,
SND_PS3_DRIVER_NAME, &the_card);
if (ret) {
pr_info("%s: request_irq failed (%d)\n", __func__, ret);
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index 7aa5b76..177f713 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -211,7 +211,7 @@
/* 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, IRQF_DISABLED,
+ au1000_dma_interrupt, 0,
&ctx->stream[s]);
set_dma_mode(ctx->stream[s].dma,
get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 3f4920d..dc8a2b2 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1419,7 +1419,7 @@
/* 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_DISABLED,
+ IRQF_TRIGGER_RISING,
codec->name, codec);
if (ret < 0) {
dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 4e3626b..ae8d680 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -268,7 +268,7 @@
nuc900_audio = nuc900_ac97_data;
if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
- IRQF_DISABLED, "nuc900-dma", substream))
+ 0, "nuc900-dma", substream))
return -EBUSY;
runtime->private_data = nuc900_audio;
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 65ea538..b5e922f 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -444,7 +444,7 @@
}
ret = request_irq(irq_res->start, s3c_ac97_irq,
- IRQF_DISABLED, "AC97", NULL);
+ 0, "AC97", NULL);
if (ret < 0) {
dev_err(&pdev->dev, "ac97: interrupt request failed.\n");
goto err4;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 916b9f9..a32fd16 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1285,7 +1285,7 @@
pm_runtime_enable(&pdev->dev);
dev_set_drvdata(&pdev->dev, master);
- ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
+ ret = request_irq(irq, &fsi_interrupt, 0,
id_entry->name, master);
if (ret) {
dev_err(&pdev->dev, "irq request err\n");
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 743d07b..a4e3f55 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -201,7 +201,7 @@
if (!drvdata->base)
return -EBUSY;
err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
- IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+ 0, dev_name(&pdev->dev), drvdata);
if (err < 0)
return err;
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index ad7d4d7..f036776 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -962,7 +962,7 @@
amd7930_idle(amd);
if (request_irq(irq, snd_amd7930_interrupt,
- IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
+ 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/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 1e3ae33..07bcfe4 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -16,6 +16,7 @@
#include <linux/firmware.h>
#include <linux/bitrev.h>
+#include <linux/kernel.h>
#include "firmware.h"
#include "chip.h"
@@ -59,21 +60,19 @@
unsigned int txt_offset; /* current position in txt_data */
};
-static u8 usb6fire_fw_ihex_nibble(const u8 n)
-{
- 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]);
+ 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;
+
*crc += val;
return val;
}
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 8beb775..3efc21c 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -67,6 +67,7 @@
* 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.
@@ -85,6 +86,7 @@
* 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/sound/usb/Makefile b/sound/usb/Makefile
index cf9ed66..ac256dc 100644
--- a/sound/usb/Makefile
+++ b/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 \
- format.o \
- endpoint.o \
- urb.o \
- pcm.o \
- helper.o \
- clock.o
+ stream.o
snd-usbmidi-lib-objs := midi.o
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493c..2cf87f5 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@
for (i = 0; i < N_URBS; i++) {
usb_kill_urb(dev->data_urbs_in[i]);
- usb_kill_urb(dev->data_urbs_out[i]);
+
+ if (test_bit(i, &dev->outurb_active_mask))
+ 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)
@@ -612,8 +616,9 @@
{
struct snd_usb_caiaq_cb_info *info = urb->context;
struct snd_usb_caiaqdev *dev;
- struct urb *out;
- int frame, len, send_it = 0, outframe = 0;
+ struct urb *out = NULL;
+ int i, frame, len, send_it = 0, outframe = 0;
+ size_t offset = 0;
if (urb->status || !info)
return;
@@ -623,7 +628,17 @@
if (!dev->streaming)
return;
- out = dev->data_urbs_out[info->index];
+ /* 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;
+ }
/* read the recently received packet and send back one which has
* the same layout */
@@ -634,7 +649,8 @@
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 = BYTES_PER_FRAME * frame;
+ out->iso_frame_desc[outframe].offset = offset;
+ offset += len;
if (len > 0) {
spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@
}
if (send_it) {
- out->number_of_packets = FRAMES_PER_URB;
+ out->number_of_packets = outframe;
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;
@@ -676,6 +696,8 @@
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)
@@ -827,6 +849,9 @@
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/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 45bc4a2..3eb605b 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -50,7 +50,8 @@
"{Native Instruments, Session I/O},"
"{Native Instruments, GuitarRig mobile}"
"{Native Instruments, Traktor Kontrol X1}"
- "{Native Instruments, Traktor Kontrol S4}");
+ "{Native Instruments, Traktor Kontrol S4}"
+ "{Native Instruments, Maschine Controller}");
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
@@ -146,6 +147,11 @@
.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/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b3101..562b0bf 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -18,6 +18,7 @@
#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
@@ -96,6 +97,7 @@
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/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7..26a121b 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@
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_5, KEY_6 };
+ KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
static unsigned short keycode_kore[] = {
KEY_FN_F1, /* "menu" */
@@ -67,6 +67,61 @@
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)
@@ -218,6 +273,29 @@
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;
}
}
@@ -400,6 +478,25 @@
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;
@@ -425,6 +522,13 @@
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:
@@ -444,6 +548,7 @@
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;
@@ -462,6 +567,7 @@
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;
}
@@ -652,6 +758,50 @@
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;
@@ -664,15 +814,17 @@
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;
}
@@ -688,4 +840,3 @@
input_unregister_device(dev->input_dev);
dev->input_dev = NULL;
}
-
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 781d9e6..c1575ea 100644
--- a/sound/usb/card.c
+++ b/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 <tiwai@suse.de>");
MODULE_DESCRIPTION("USB Audio");
@@ -185,7 +185,7 @@
return -EINVAL;
}
- if (! snd_usb_parse_audio_endpoints(chip, interface)) {
+ if (! snd_usb_parse_audio_interface(chip, interface)) {
usb_set_interface(dev, interface, 0); /* reset the current interface */
usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
return -EINVAL;
@@ -530,8 +530,11 @@
return chip;
__error:
- if (chip && !chip->num_interfaces)
- snd_card_free(chip->card);
+ if (chip) {
+ if (!chip->num_interfaces)
+ snd_card_free(chip->card);
+ chip->probing = 0;
+ }
mutex_unlock(®ister_mutex);
__err_val:
return NULL;
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ae4251d..a39edcc 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -94,6 +94,8 @@
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/sound/usb/clock.c b/sound/usb/clock.c
index 075195e..379baad 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -91,7 +91,7 @@
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), 1000);
+ &buf, sizeof(buf));
if (ret < 0)
return ret;
@@ -118,7 +118,7 @@
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), 1000);
+ &data, sizeof(data));
if (err < 0) {
snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
@@ -222,7 +222,7 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 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 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 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 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 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 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
return err;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7c0d21e..81c6ede 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -15,436 +15,951 @@
*
*/
+#include <linux/gfp.h>
#include <linux/init.h>
-#include <linux/slab.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
-#include <linux/usb/audio-v2.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include "usbaudio.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"
+#include "card.h"
+#include "endpoint.h"
+#include "pcm.h"
/*
- * free a substream
+ * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
+ * this will overflow at approx 524 kHz
*/
-static void free_substream(struct snd_usb_substream *subs)
+static inline unsigned get_usb_full_speed_rate(unsigned int rate)
{
- 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);
+ return ((rate << 13) + 62) / 125;
}
-
/*
- * free a usb stream instance
+ * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
+ * this will overflow at approx 4 MHz
*/
-static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
+static inline unsigned get_usb_high_speed_rate(unsigned int rate)
{
- free_substream(&stream->substream[0]);
- free_substream(&stream->substream[1]);
- list_del(&stream->list);
- kfree(stream);
+ return ((rate << 10) + 62) / 125;
}
-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.
+ * unlink active urbs.
*/
-int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp)
+static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
{
- struct list_head *p;
- struct snd_usb_stream *as;
- struct snd_usb_substream *subs;
- struct snd_pcm *pcm;
- int err;
+ struct snd_usb_audio *chip = subs->stream->chip;
+ unsigned int i;
+ int async;
- 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;
+ 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);
+ }
}
}
- /* 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) {
+ 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;
+}
- /* 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;
+
+/*
+ * 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;
}
- 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);
+}
+
+/*
+ * 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
- strcpy(pcm->name, "USB Audio");
+ 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;
- snd_usb_init_substream(as, stream, fp);
+ if (subs->fill_max)
+ subs->curpacksize = subs->maxpacksize;
+ else
+ subs->curpacksize = maxsize;
- list_add(&as->list, &chip->pcm_list);
- chip->pcm_devs++;
+ if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
+ packs_per_ms = 8 >> subs->datainterval;
+ else
+ packs_per_ms = 1;
- snd_usb_proc_pcm_format_add(as);
+ 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;
}
-static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
- struct usb_host_interface *alts,
- int protocol, int iface_no)
+/* determine the number of frames in the next packet */
+static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
{
- /* 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;
+ if (subs->fill_max)
+ return subs->maxframesize;
+ else {
+ subs->phase = (subs->phase & 0xffff)
+ + (subs->freqm << subs->datainterval);
+ return min(subs->phase >> 16, subs->maxframesize);
+ }
+}
- csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+/*
+ * 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;
- /* 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);
+ 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;
+}
- 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);
+/*
+ * 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;
+}
+
+/*
+ * 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;
+ int est_delay;
+
+ spin_lock_irqsave(&subs->lock, flags);
+
+ 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;
+
+ /*
+ * 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.
+ */
+ 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;
+}
+
+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;
}
- 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;
+ return -EINVAL;
}
-static struct uac2_input_terminal_descriptor *
- snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
- int terminal_id)
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct uac2_input_terminal_descriptor *term = NULL;
+ struct snd_usb_substream *subs = substream->runtime->private_data;
- while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
- ctrl_iface->extralen,
- term, UAC_INPUT_TERMINAL))) {
- if (term->bTerminalID == terminal_id)
- return term;
+ 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 NULL;
+ return -EINVAL;
}
-static struct uac2_output_terminal_descriptor *
- snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
- int terminal_id)
+int snd_usb_substream_prepare(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime)
{
- struct uac2_output_terminal_descriptor *term = NULL;
+ /* clear urbs (to be sure) */
+ deactivate_urbs(subs, 0, 1);
+ wait_clear_urbs(subs);
- while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
- ctrl_iface->extralen,
- term, UAC_OUTPUT_TERMINAL))) {
- if (term->bTerminalID == terminal_id)
- return term;
+ /* 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 NULL;
-}
-
-int snd_usb_parse_audio_endpoints(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_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/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 64dd0db..88eb63a 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -1,11 +1,21 @@
#ifndef __USBAUDIO_ENDPOINT_H
#define __USBAUDIO_ENDPOINT_H
-int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip,
- int iface_no);
+void snd_usb_init_substream(struct snd_usb_stream *as,
+ int stream,
+ struct audioformat *fp);
-int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip,
- 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_ENDPOINT_H */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 8d042dc..89421d1 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -286,7 +286,7 @@
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), 1000);
+ tmp, sizeof(tmp));
if (ret < 0) {
snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n",
@@ -307,7 +307,7 @@
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, 1000);
+ data, data_size);
if (ret < 0) {
snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n",
diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index f280c19..9eed8f4 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -81,7 +81,7 @@
*/
int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
__u8 requesttype, __u16 value, __u16 index, void *data,
- __u16 size, int timeout)
+ __u16 size)
{
int err;
void *buf = NULL;
@@ -92,7 +92,7 @@
return -ENOMEM;
}
err = usb_control_msg(dev, pipe, request, requesttype,
- value, index, buf, size, timeout);
+ value, index, buf, size, 1000);
if (size > 0) {
memcpy(data, buf, size);
kfree(buf);
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index 09bd943..805c300 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -8,7 +8,7 @@
int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe,
__u8 request, __u8 requesttype, __u16 value, __u16 index,
- void *data, __u16 size, int timeout);
+ void *data, __u16 size);
unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
struct usb_host_interface *alts);
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index f928910..e21f026 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -816,6 +816,22 @@
.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)
{
@@ -2163,6 +2179,17 @@
/* 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/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76..60f65ac 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@
if (p && p->dB) {
cval->dBmin = p->dB->min;
cval->dBmax = p->dB->max;
+ cval->initialized = 1;
}
}
@@ -295,7 +296,7 @@
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, 100) >= val_len) {
+ buf, val_len) >= val_len) {
*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
snd_usb_autosuspend(cval->mixer->chip);
return 0;
@@ -332,7 +333,7 @@
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, 1000);
+ buf, size);
snd_usb_autosuspend(chip);
if (ret < 0) {
@@ -444,7 +445,7 @@
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, 100) >= 0) {
+ buf, val_len) >= 0) {
snd_usb_autosuspend(chip);
return 0;
}
@@ -880,8 +881,17 @@
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
} else {
- if (! cval->initialized)
- get_min_max(cval, 0);
+ 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);
+ }
+ }
uinfo->value.integer.min = 0;
uinfo->value.integer.max =
(cval->max - cval->min + cval->res - 1) / cval->res;
@@ -1092,7 +1102,7 @@
" Switch" : " Volume");
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
- if (cval->dBmin < cval->dBmax) {
+ if (cval->dBmin < cval->dBmax || !cval->initialized) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1191,6 +1201,11 @@
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 {
@@ -1244,7 +1259,7 @@
build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0);
}
} else { /* UAC_VERSION_2 */
- for (i = 0; i < 30/2; i++) {
+ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) {
unsigned int ch_bits = 0;
unsigned int ch_read_only = 0;
@@ -1934,15 +1949,13 @@
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 = hostif->extra;
- state.buflen = hostif->extralen;
+ state.buffer = mixer->hostif->extra;
+ state.buflen = mixer->hostif->extralen;
/* check the mapping table */
for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1968,8 @@
}
p = NULL;
- while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+ while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+ p, UAC_OUTPUT_TERMINAL)) != NULL) {
if (mixer->protocol == UAC_VERSION_1) {
struct uac1_output_terminal_descriptor *desc = p;
@@ -2162,17 +2176,15 @@
/* 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(hostif)->bNumEndpoints < 1)
+ if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
return 0;
- ep = get_endpoint(hostif, 0);
+ ep = get_endpoint(mixer->hostif, 0);
if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
return 0;
@@ -2202,7 +2214,6 @@
};
struct usb_mixer_interface *mixer;
struct snd_info_entry *entry;
- struct usb_host_interface *host_iface;
int err;
strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2230,8 @@
return -ENOMEM;
}
- host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+ switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
case UAC_VERSION_1:
default:
mixer->protocol = UAC_VERSION_1;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14d..81b2d8a 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
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/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 3d0f487..ab125ee 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -190,18 +190,18 @@
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, 100);
+ !value, 0, NULL, 0);
/* 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, 100);
+ !value, 0, NULL, 0);
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, 100);
+ value, index + 2, NULL, 0);
if (err < 0)
return err;
mixer->audigy2nx_leds[index] = value;
@@ -299,7 +299,7 @@
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, 100);
+ jacks[i].unitid << 8, buf, 3);
if (err == 3 && (buf[0] == 3 || buf[0] == 6))
snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]);
else
@@ -332,7 +332,7 @@
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, 100);
+ 50, 0, &new_status, 1);
if (err < 0)
return err;
mixer->xonar_u1_status = new_status;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index b8dcbf4..0220b0f 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -28,12 +28,36 @@
#include "card.h"
#include "quirks.h"
#include "debug.h"
-#include "urb.h"
+#include "endpoint.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.
*/
@@ -45,6 +69,8 @@
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);
}
@@ -126,7 +152,7 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n",
dev->devnum, iface, ep);
return err;
@@ -150,7 +176,7 @@
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), 1000)) < 0) {
+ data, sizeof(data))) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
dev->devnum, iface, fmt->altsetting);
return err;
@@ -417,6 +443,8 @@
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/sound/usb/pcm.h b/sound/usb/pcm.h
index ed3e283..df7a003 100644
--- a/sound/usb/pcm.h
+++ b/sound/usb/pcm.h
@@ -1,6 +1,9 @@
#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/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index dba0b7f..b61945f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -39,6 +39,17 @@
.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),
@@ -1678,6 +1689,20 @@
}
},
{
+ /* 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) {
/* .vendor_name = "BOSS", */
@@ -1707,6 +1732,40 @@
}
}
},
+{
+ 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 */
{
@@ -2417,6 +2476,12 @@
.idProduct = 0x1020,
},
+/* KeithMcMillen Stringport */
+{
+ USB_DEVICE(0x1f38, 0x0001),
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+
/* Miditech devices */
{
USB_DEVICE(0x4752, 0x0011),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 77762c9..2e5bc73 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -34,6 +34,7 @@
#include "endpoint.h"
#include "pcm.h"
#include "clock.h"
+#include "stream.h"
/*
* handle the quirks for the contained interfaces
@@ -106,7 +107,7 @@
alts = &iface->altsetting[0];
altsd = get_iface_desc(alts);
- err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber);
+ err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber);
if (err < 0) {
snd_printk(KERN_ERR "cannot setup if %d: error %d\n",
altsd->bInterfaceNumber, err);
@@ -147,7 +148,7 @@
stream = (fp->endpoint & USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- err = snd_usb_add_audio_endpoint(chip, stream, fp);
+ err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
kfree(fp);
kfree(rate_table);
@@ -254,7 +255,7 @@
stream = (fp->endpoint & USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- err = snd_usb_add_audio_endpoint(chip, stream, fp);
+ err = snd_usb_add_audio_stream(chip, stream, fp);
if (err < 0) {
kfree(fp);
return err;
@@ -306,6 +307,7 @@
[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,
@@ -338,7 +340,7 @@
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, 1000);
+ 0x10, 0x43, 0x0001, 0x000a, NULL, 0);
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));
@@ -359,11 +361,11 @@
snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a,
USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- 0, 0, &buf, 1, 1000);
+ 0, 0, &buf, 1);
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, 1000);
+ 1, 2000, NULL, 0);
return -ENODEV;
}
return 0;
@@ -406,7 +408,7 @@
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, 1000);
+ 0, 0, &buf, 4);
}
static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
@@ -426,7 +428,7 @@
*/
static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
{
- int err, reg;
+ int err = 0, reg;
int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
new file mode 100644
index 0000000..5ff8010
--- /dev/null
+++ b/sound/usb/stream.c
@@ -0,0 +1,452 @@
+/*
+ * 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 <linux/init.h>
+#include <linux/slab.h>
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#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/sound/usb/stream.h b/sound/usb/stream.h
new file mode 100644
index 0000000..c97f679
--- /dev/null
+++ b/sound/usb/stream.h
@@ -0,0 +1,12 @@
+#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/sound/usb/urb.c b/sound/usb/urb.c
deleted file mode 100644
index e184349..0000000
--- a/sound/usb/urb.c
+++ /dev/null
@@ -1,941 +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 <linux/gfp.h>
-#include <linux/init.h>
-#include <linux/usb.h>
-#include <linux/usb/audio.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-
-#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/sound/usb/urb.h b/sound/usb/urb.h
deleted file mode 100644
index 888da38..0000000
--- a/sound/usb/urb.h
+++ /dev/null
@@ -1,21 +0,0 @@
-#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/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 1e79986..3e2b035 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -80,6 +80,7 @@
QUIRK_MIDI_CME,
QUIRK_MIDI_AKAI,
QUIRK_MIDI_US122L,
+ QUIRK_MIDI_FTDI,
QUIRK_AUDIO_STANDARD_INTERFACE,
QUIRK_AUDIO_FIXED_ENDPOINT,
QUIRK_AUDIO_EDIROL_UAXX,