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# SPDX-License-Identifier: GPL-2.0-only
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# ALSA soundcard-configuration
config SND_TIMER
tristate
config SND_PCM
tristate
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select SND_TIMER if SND_PCM_TIMER
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config SND_PCM_ELD
bool
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config SND_PCM_IEC958
bool
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config SND_DMAENGINE_PCM
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tristate
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config SND_HWDEP
tristate
ALSA: seq: Allow the modular sequencer registration
Many drivers bind the sequencer stuff in off-load by another driver
module, so that it's loaded only on demand. In the current code, this
mechanism doesn't work when the driver is built-in while the sequencer
is module. We check with IS_REACHABLE() and enable only when the
sequencer is in the same level of build.
However, this is basically a overshoot. The binder code
(snd-seq-device) is an individual module from the sequencer core
(snd-seq), and we just have to make the former a built-in while
keeping the latter a module for allowing the scenario like the above.
This patch achieves that by rewriting Kconfig slightly. Now, a driver
that provides the manual sequencer device binding should select
CONFIG_SND_SEQ_DEVICE in a way as
select SND_SEQ_DEVICE if SND_SEQUENCER != n
Note that the "!=n" is needed here to avoid the influence of the
sequencer core is module while the driver is built-in.
Also, since rawmidi.o may be linked with snd_seq_device.o when
built-in, we have to shuffle the code to make the linker happy.
(the kernel linker isn't smart enough yet to handle such a case.)
That is, snd_seq_device.c is moved to sound/core from sound/core/seq,
as well as Makefile.
Last but not least, the patch replaces the code using IS_REACHABLE()
with IS_ENABLED(), since now the condition meets always when enabled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-06-09 16:11:58 +03:00
config SND_SEQ_DEVICE
tristate
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config SND_RAWMIDI
tristate
ALSA: seq: Allow the modular sequencer registration
Many drivers bind the sequencer stuff in off-load by another driver
module, so that it's loaded only on demand. In the current code, this
mechanism doesn't work when the driver is built-in while the sequencer
is module. We check with IS_REACHABLE() and enable only when the
sequencer is in the same level of build.
However, this is basically a overshoot. The binder code
(snd-seq-device) is an individual module from the sequencer core
(snd-seq), and we just have to make the former a built-in while
keeping the latter a module for allowing the scenario like the above.
This patch achieves that by rewriting Kconfig slightly. Now, a driver
that provides the manual sequencer device binding should select
CONFIG_SND_SEQ_DEVICE in a way as
select SND_SEQ_DEVICE if SND_SEQUENCER != n
Note that the "!=n" is needed here to avoid the influence of the
sequencer core is module while the driver is built-in.
Also, since rawmidi.o may be linked with snd_seq_device.o when
built-in, we have to shuffle the code to make the linker happy.
(the kernel linker isn't smart enough yet to handle such a case.)
That is, snd_seq_device.c is moved to sound/core from sound/core/seq,
as well as Makefile.
Last but not least, the patch replaces the code using IS_REACHABLE()
with IS_ENABLED(), since now the condition meets always when enabled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-06-09 16:11:58 +03:00
select SND_SEQ_DEVICE if SND_SEQUENCER != n
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config SND_COMPRESS_OFFLOAD
tristate
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config SND_JACK
bool
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# enable input device support in jack layer
config SND_JACK_INPUT_DEV
bool
depends on SND_JACK
default y if INPUT=y || INPUT=SND
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config SND_OSSEMUL
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bool "Enable OSS Emulation"
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select SOUND_OSS_CORE
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help
This option enables the build of OSS emulation layer.
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config SND_MIXER_OSS
tristate "OSS Mixer API"
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depends on SND_OSSEMUL
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help
To enable OSS mixer API emulation (/dev/mixer*), say Y here
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and read <file:Documentation/sound/designs/oss-emulation.rst>.
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Many programs still use the OSS API, so say Y.
To compile this driver as a module, choose M here: the module
will be called snd-mixer-oss.
config SND_PCM_OSS
tristate "OSS PCM (digital audio) API"
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depends on SND_OSSEMUL
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select SND_PCM
help
To enable OSS digital audio (PCM) emulation (/dev/dsp*), say Y
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here and read <file:Documentation/sound/designs/oss-emulation.rst>.
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Many programs still use the OSS API, so say Y.
To compile this driver as a module, choose M here: the module
will be called snd-pcm-oss.
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config SND_PCM_OSS_PLUGINS
bool "OSS PCM (digital audio) API - Include plugin system"
depends on SND_PCM_OSS
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default y
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help
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If you disable this option, the ALSA's OSS PCM API will not
support conversion of channels, formats and rates. It will
behave like most of new OSS/Free drivers in 2.4/2.6 kernels.
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config SND_PCM_TIMER
bool "PCM timer interface" if EXPERT
default y
help
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If you disable this option, pcm timer will be unavailable, so
those stubs that use pcm timer (e.g. dmix, dsnoop & co) may work
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incorrectlly.
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For some embedded devices, we may disable it to reduce memory
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footprint, about 20KB on x86_64 platform.
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config SND_HRTIMER
tristate "HR-timer backend support"
depends on HIGH_RES_TIMERS
select SND_TIMER
help
Say Y here to enable HR-timer backend for ALSA timer. ALSA uses
the hrtimer as a precise timing source. The ALSA sequencer code
also can use this timing source.
To compile this driver as a module, choose M here: the module
will be called snd-hrtimer.
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config SND_DYNAMIC_MINORS
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bool "Dynamic device file minor numbers"
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help
If you say Y here, the minor numbers of ALSA device files in
/dev/snd/ are allocated dynamically. This allows you to have
more than 8 sound cards, but requires a dynamic device file
system like udev.
If you are unsure about this, say N here.
ALSA: Add kconfig to specify the max card numbers
Currently ALSA supports up to 32 card instances when the dynamic minor
is used. While 32 cards are usually big enough for normal use cases,
there are sometimes weird requirements with more card support.
Actually, this limitation, 32, comes from the index option, where you
can pass the bit mask to assign the card. Other than that, we can
actually give more cards up to the minor number limits (currently 256,
which can be extended more, too).
This patch adds a new Kconfig to specify the max card numbers, and
changes a few places to accept more than 32 cards.
The only incompatibility with high card numbers would be the handling
of index option. The index option can be still used to pass the
bitmask for card assignments, but this works only up to 32 slots.
More than 32, no bitmask style option is available but only a single
slot can be specified via index option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-15 10:46:39 +04:00
config SND_MAX_CARDS
int "Max number of sound cards"
range 4 256
default 32
depends on SND_DYNAMIC_MINORS
help
Specify the max number of sound cards that can be assigned
on a single machine.
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config SND_SUPPORT_OLD_API
bool "Support old ALSA API"
default y
help
Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3
or older).
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config SND_PROC_FS
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bool "Sound Proc FS Support" if EXPERT
depends on PROC_FS
default y
help
Say 'N' to disable Sound proc FS, which may reduce code size about
9KB on x86_64 platform.
If unsure say Y.
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config SND_VERBOSE_PROCFS
bool "Verbose procfs contents"
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depends on SND_PROC_FS
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default y
help
Say Y here to include code for verbose procfs contents (provides
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useful information to developers when a problem occurs). On the
other side, it makes the ALSA subsystem larger.
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config SND_VERBOSE_PRINTK
bool "Verbose printk"
help
Say Y here to enable verbose log messages. These messages
will help to identify source file and position containing
printed messages.
You don't need this unless you're debugging ALSA.
config SND_DEBUG
bool "Debug"
help
Say Y here to enable ALSA debug code.
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config SND_DEBUG_VERBOSE
bool "More verbose debug"
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depends on SND_DEBUG
help
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Say Y here to enable extra-verbose debugging messages.
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Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages.
So, say Y only if you are ready to be annoyed.
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config SND_PCM_XRUN_DEBUG
bool "Enable PCM ring buffer overrun/underrun debugging"
default n
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depends on SND_DEBUG && SND_VERBOSE_PROCFS
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help
Say Y to enable the PCM ring buffer overrun/underrun debugging.
It is usually not required, but if you have trouble with
sound clicking when system is loaded, it may help to determine
the process or driver which causes the scheduling gaps.
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ALSA: control: Add verification for kctl accesses
The current implementation of ALSA control API fully relies on the
callbacks of each driver, and there is no verification of the values
passed via API. This patch is an attempt to improve the situation
slightly by adding the validation code for the values stored via info
and get callbacks.
The patch adds a new kconfig, CONFIG_SND_CTL_VALIDATION. It depends
on CONFIG_SND_DEBUG and off as default since the validation would
require a slight overhead including the additional call of info
callback at each get callback invocation.
When this config is enabled, the values stored by each info callback
invocation are verified, namely:
- Whether the info type is valid
- Whether the number of enum items is non-zero
- Whether the given info count is within the allowed boundary
Similarly, the values stored at each get callback are verified as
well:
- Whether the values are within the given range
- Whether the values are aligned with the given step
- Whether any further changes are seen in the data array over the
given info count
The last point helps identifying a possibly invalid data type access,
typically a case where the info callback declares the type being
SNDRV_CTL_ELEM_TYPE_ENUMERATED while the get/put callbacks store
the values in value.integer.value[] array.
When a validation fails, the ALSA core logs an error message including
the device and the control ID, and the API call also returns an
error. So, with the new validation turned on, the driver behavior
difference may be visible on user-space, too -- it's intentional,
though, so that we can catch an error more clearly.
The patch also introduces a new ctl access type,
SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK. A driver may pass this flag with
other access bits to indicate that the ctl element won't be verified.
It's useful when a driver code is specially written to access the data
greater than info->count size by some reason. For example, this flag
is actually set now in HD-audio HDMI codec driver which needs to clear
the data array in the case of the disconnected monitor.
Also, the PCM channel-map helper code is slightly modified to avoid
the false-positive hit by this validation code, too.
Link: https://lore.kernel.org/r/20200104083556.27789-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-01-04 11:35:56 +03:00
config SND_CTL_VALIDATION
bool "Perform sanity-checks for each control element access"
depends on SND_DEBUG
help
Say Y to enable the additional validation of each control element
access, including sanity-checks like whether the values returned
from the driver are in the proper ranges or the check of the invalid
access at out-of-array areas.
ALSA: jack: implement software jack injection via debugfs
This change adds audio jack injection feature through debugfs, with
this feature, we could validate alsa userspace changes by injecting
plugin or plugout events to the non-phantom audio jacks.
With this change, the sound core will build the folders
$debugfs_mount_dir/sound/cardN if SND_DEBUG and DEBUG_FS are enabled.
And if users also enable the SND_JACK_INJECTION_DEBUG, the jack
injection nodes will be built in the folder cardN like below:
$tree $debugfs_mount_dir/sound
$debugfs_mount_dir/sound
├── card0
│ ├── HDMI_DP_pcm_10_Jack
│ │ ├── jackin_inject
│ │ ├── kctl_id
│ │ ├── mask_bits
│ │ ├── status
│ │ ├── sw_inject_enable
│ │ └── type
...
│ └── HDMI_DP_pcm_9_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── card1
├── HDMI_DP_pcm_5_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
...
├── Headphone_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── Headset_Mic_Jack
├── jackin_inject
├── kctl_id
├── mask_bits
├── status
├── sw_inject_enable
└── type
The nodes kctl_id, mask_bits, status and type are read-only, users
could check jack or jack_kctl's information through them.
The nodes sw_inject_enable and jackin_inject are directly used for
injection. The sw_inject_enable is read-write, users could check if
software injection is enabled or not on this jack, and users could
echo 1 or 0 to enable or disable software injection on this jack. Once
the injection is enabled, the jack will not change by hardware events
anymore, once the injection is disabled, the jack will restore the
last reported hardware events to the jack. The jackin_inject is
write-only, if the injection is enabled, users could echo 1 or 0 to
this node to inject plugin or plugout events to this jack.
For the detailed usage information on these nodes, please refer to
Documentation/sound/designs/jack-injection.rst.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210127085639.74954-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-27 11:56:39 +03:00
config SND_JACK_INJECTION_DEBUG
bool "Sound jack injection interface via debugfs"
depends on SND_JACK && SND_DEBUG && DEBUG_FS
help
This option can be used to enable or disable sound jack
software injection.
Say Y if you are debugging via jack injection interface.
If unsure select "N".
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config SND_VMASTER
bool
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config SND_DMA_SGBUF
def_bool y
depends on X86
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config SND_CTL_LED
tristate
select NEW_LEDS if SND_CTL_LED
select LEDS_TRIGGERS if SND_CTL_LED
select LEDS_TRIGGER_AUDIO if SND_CTL_LED
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source "sound/core/seq/Kconfig"
