USB-audio driver assumes that the normal resume would preserve the
device configuration while reset_resume wouldn't, and tries to restore
the mixer elements only at reset_resume callback. However, this seems
too naive, and some devices do behave differently, resetting the
volume at the normal resume; this resulted in the inconsistent volume
that surprised users.
This patch changes the mixer resume code to handle both the normal and
reset resume in the same way, always restoring the original mixer
element values. This allows us to unify the both callbacks as well as
dropping the no longer used reset_resume field, which ends up with a
good code reduction.
A slight behavior change by this patch is that now we assign
restore_mixer_value() as the default resume callback, and the function
is no longer called at reset-resume when the resume callback is
overridden by the quirk function. That is, if needed, the quirk
resume function would have to handle similarly as
restore_mixer_value() by itself.
Reported-by: En-Shuo Hsu <enshuo@chromium.org>
Cc: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsMew@mail.gmail.com
Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another device ID for JBL Quantum 800. It requires the same quirk as
other JBL Quantum devices.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210831002531.116957-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For making user to switch back to the old playback mode, this patch
adds a new module option 'lowlatency' to snd-usb-audio driver.
When user face a regression due to the recent low-latency playback
support, they can test easily by passing lowlatency=0 option without
rebuilding the kernel.
Fixes: 307cc9baac ("ALSA: usb-audio: Reduce latency at playback start, take#2")
Link: https://lore.kernel.org/r/20210829073830.22686-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change for low latency playback works in most of test cases
but it turned out still to hit errors on some use cases, most notably
with JACK with small buffer sizes. This is because USB-audio driver
fills up and submits full URBs at the beginning, while the URBs would
return immediately and try to fill more -- that can easily trigger
XRUN. It was more or less expected, but in the small buffer size, the
problem became pretty obvious.
Fixing this behavior properly would require the change of the
fundamental driver design, so it's no trivial task, unfortunately.
Instead, here we work around the problem just by switching back to the
old method when the given configuration is too fragile with the low
latency stream handling. As a threshold, we calculate the total
buffer bytes in all plus one URBs, and check whether it's beyond the
PCM buffer bytes. The one extra URB is needed because XRUN happens at
the next submission after the first round.
Fixes: 307cc9baac ("ALSA: usb-audio: Reduce latency at playback start, take#2")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210827203311.5987-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent quirk for WALKMAN (commit 7af5a14371c1: "ALSA: usb-audio:
Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other
devices and is worth to be put into the common quirk flags.
This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a
quirk table entry for the device.
Link: https://lore.kernel.org/r/20210824055720.9240-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a regression report for USB-audio with Sony WALKMAN NW-A45
DAC device where no sound is audible on recent kernel. The bisection
resulted in the code change wrt endpoint management, and the further
debug session revealed that it was caused by the order of the USB
audio interface. In the earlier code, we always set up the USB
interface at first before other setups, but it was changed to be done
at the last for UAC2/3, which is more standard way, while keeping the
old way for UAC1. OTOH, this device seems requiring the setup of the
interface at first just like UAC1.
This patch works around the regression by applying the interface setup
specifically for the WALKMAN at the beginning of the endpoint setup
procedure. This change is written straightforwardly to be easily
backported in old kernels. A further cleanup to move the workaround
into a generic quirk section will follow in a later patch.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214105
Link: https://lore.kernel.org/r/20210824054700.8236-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a second mixer control to Digidesign Mbox
to select between Analog/SPDIF capture.
Users will note that selecting the SPDIF input source
automatically switches the clock mode to sync to SPDIF,
which is a feature of the hardware.
(You can change the clock source back to internal if you want
to capture from spdif but not sync to its clock although this mode
is probably not recommended).
Unfortunately, starting the stream resets both modes
to Internally clocked and Analog input mode.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Tested-by: Damien Zammit <damien@zamaudio.com>
Link: https://lore.kernel.org/r/20210813113402.11849-1-damien@zamaudio.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't populate array names_to_check on the stack but instead it
static. Makes the object code smaller by 56 bytes. Also clean
up checkpatch warning by adding extra const for names_to_check
and pointer s.
Before:
text data bss dec hex filename
103512 34380 0 137892 21aa4 ./sound/usb/mixer.o
After:
text data bss dec hex filename
103264 34572 0 137836 21a6c ./sound/usb/mixer.o
(gcc version 10.2.0)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210803122839.7143-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new module option, quirk_flags, for allowing user to
try some additional device-specific quirk behavior more easily.
When this option is set to non-zero, it overrides the quirk_flags, and
the specific workaround is applied.
Link: https://lore.kernel.org/r/20210729074404.19728-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer code has a flag ignore_ctl_error for ignoring the errors
returned from the device wrt mixer accesses, and this is set from the
entries in mixer_maps.c, as well as ignore_ctl_error module option.
Those can be well integrated into the new quirk_flags field, too.
Link: https://lore.kernel.org/r/20210729074404.19728-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-suspend suppression workaround for Lenovo machines are
handled in quirks-table.h. Now it's more easier to handle with
quirk_flags.
Link: https://lore.kernel.org/r/20210729074404.19728-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The rate validation at the device probe is applied only to the
specific devices (currently only for MOTU devices), and this check can
be moved to quirk_flags gracefully, too.
Link: https://lore.kernel.org/r/20210729074404.19728-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We apply some delay for the control messages on certain devices as a
workaround, and this can be moved into the quirk_flags as well.
Currently there are three different delay periods (1ms, 5ms and 20ms),
so three different quirk bits are assigned for them.
Link: https://lore.kernel.org/r/20210729073855.19043-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is another quirk for the transfer, and that's currently specific
to Zoom R16/24, handled in create_standard_audio_quirk(). Let's move
this also to the new quirk_flags.
Link: https://lore.kernel.org/r/20210729073855.19043-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The txfr_quirk field was meant for aligning the transfer, and it's set
for certain devices in quirks-table.h. Now we can move that stuff
also to the new quirk_flags gracefully, and reduce the quirks-table.h
entries (that are exposed to module device table).
As the quirks-table.h entries are also with the name string override,
provide the corresponding entries to the usb_audio_names[] table,
too.
Link: https://lore.kernel.org/r/20210729073855.19043-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices that can have media-controller API entries are currently
specified via tables in quirks-table.h, as a part of descriptor
override. This can fit better to the new quirk_flags, as we just need
a matching with the given ID and create the MC entries accordingly.
Link: https://lore.kernel.org/r/20210729073855.19043-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As more and more device-specific workarounds came up and gathered in
various places, it becomes harder to manage. Now it's time to clean
up and collect workarounds more consistently and make them more easily
applicable.
This patch is the first step for that: a new field quirk_flags is
introduced in snd_usb_audio struct to contain the bit flags for
various device-specific quirks. Those are separate one from the
quirks in quirks-table.h; the quirks-table.h entries are for more
intrusive stuff that needs the descriptor override, while the new
quirk_flags is for easier ones that are tied with the vendor:product
IDs.
In this patch, as the first example, we convert the list of devices
and vendors to ignore GET_SAMPLE_RATE, formerly defined in
snb_usb_get_sample_rate_quirk().
Link: https://lore.kernel.org/r/20210729073855.19043-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the resume on Lenovo machines seems causing a
regression on others. It's because the change always triggers the
connector selection no matter which widget node type is.
This patch addresses the regression by setting the resume callback
selectively only for the connector widget.
Fixes: 44609fc01f ("ALSA: usb-audio: Check connector value on resume")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897
Link: https://lore.kernel.org/r/20210729185126.24432-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apparently JBL Quantum 600 has multiple hardware revisions. Apply
registration quirk to another device id as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727093326.1153366-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The change to restore the autosuspend from the disabled state uses a
wrong check: namely, it should have been the exact comparison of the
quirk_type instead of the bitwise and (&). Otherwise it matches
wrongly with the other quirk types.
Although re-enabling the autosuspend for the already enabled device
shouldn't matter much, it's better to fix the unbalanced call.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hr1flh9ov.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The following scenario describes an echo test for
Samsung USBC Headset (AKG) with VID/PID (0x04e8/0xa051).
We first start a capture stream(USB IN transfer) in 96Khz/24bit/1ch mode.
In clock find source function, we get value 0x2 for clock selector
and 0x1 for clock source.
Kernel-4.14 behavior
Since clock source is valid so clock selector was not set again.
We pass through this function and start a playback stream(USB OUT transfer)
in 48Khz/32bit/2ch mode. This time we get value 0x1 for clock selector
and 0x1 for clock source. Finally clock id with this setting is 0x9.
Kernel-5.10 behavior
Clock selector was always set one more time even it is valid.
When we start a playback stream, we will get 0x2 for clock selector
and 0x1 for clock source. In this case clock id becomes 0xA.
This is an incorrect clock source setting and results in severe noises.
We see wrong data rate in USB IN transfer.
(From 288 bytes/ms becomes 144 bytes/ms) It should keep in 288 bytes/ms.
This earphone works fine on older kernel version load because
this is a newly-added behavior.
Fixes: d2e8f64125 ("ALSA: usb-audio: Explicitly set up the clock selector")
Signed-off-by: chihhao.chen <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/1627100621-19225-1-git-send-email-chihhao.chen@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The values of the line output controls can change when the SW/HW
switches are set to HW, and also when speaker switching is enabled.
These notifications were sent with a mask of only
SNDRV_CTL_EVENT_MASK_INFO. Change the notifications to set the
SNDRV_CTL_EVENT_MASK_VALUE mask bit as well.
When the mute control is updated, the notification was sent with a
mask of SNDRV_CTL_EVENT_MASK_INFO. Change the mask to the correct
value of SNDRV_CTL_EVENT_MASK_VALUE.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/8192e15ba62fa4bc90425c005f265c0de530be20.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the hardware mute button is pressed, private->vol_updated is set
so that the mute status is invalidated. As the channel mute values may
be affected by the global mute value, update scarlett2_mute_ctl_get()
to call scarlett2_update_volumes() if private->vol_updated is set.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/aa18ddbf8d8bd7f31832ab1b6b6057c00b931202.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These devices has two interfaces, but only the second interface
contains the capture endpoint, thus quirk is required to delay the
registration until the second interface appears.
Tested-by: Jakub Fišer <jakub@ufiseru.cz>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210721235605.53741-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we've added a new usb_mixer element type, USB_MIXER_BESPOKEN,
but it wasn't added in the table in snd_usb_mixer_dump_cval(). This
is no big problem since each bespoken type should have its own dump
method, but it still isn't disallowed to use the standard one, so we
should cover it as well. Along with it, define the table with the
explicit array initializer for avoiding other pitfalls.
Fixes: 785b6f29a7 ("ALSA: usb-audio: scarlett2: Fix wrong resume call")
Reported-by: Pavel Machek <pavel@denx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210714084836.1977-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another attempt for the reduction of the latency at the start
of a USB audio playback stream. The first attempt in the commit
9ce650a75a caused an unexpected regression (a deadlock with pipewire
usage) and was later reverted by the commit 4b820e167b. The devils
are always living in details, of course; the cause of the deadlock was
the call of snd_pcm_period_elapsed() inside prepare_playback_urb()
callback. In the original code, this callback is never called from
the stream lock context as it's driven solely from the URB complete
callback. Along with the movement of the URB submission into the
trigger START, this prepare call may be also executed in the stream
lock context, hence it deadlocked with the another lock in
snd_pcm_period_elapsed(). (Note that this happens only conditionally
with a small period size that matches with the URB buffer length,
which was a reason I overlooked during my tests. Also, the problem
wasn't seen in the capture stream because the capture stream handles
the period-elapsed only at retire callback that isn't executed at the
trigger.)
If it were only about avoiding the deadlock, it'd be possible to use
snd_pcm_period_elapsed_under_stream_lock() as a solution. However, in
general, the period elapsed notification must be sent after the actual
stream start, and replacing the call wouldn't satisfy the pattern.
A better option is to delay the notification after the stream start
procedure finished, instead. In the case of USB framework, one of the
fitting place would be the complete callback of the first URB.
So, as a workaround of the deadlock and the order fixes above, in
addition to the re-applying the changes in the commit 9ce650a75a,
this patch introduces a new flag indicating the delayed period-elapsed
handling and sets it under the possible deadlock condition
(i.e. prepare callback being called before subs->running is set).
Once when the flag is set, the period-elapsed call is handled at a
later URB complete call instead.
As a reference for the original motivation for the low-latency change,
I cite here again:
| USB-audio driver behaves a bit strangely for the playback stream --
| namely, it starts sending silent packets at PCM prepare state while
| the actual data is submitted at first when the trigger START is
| kicked off. This is a workaround for the behavior where URBs are
| processed too quickly at the beginning. That is, if we start
| submitting URBs at trigger START, the first few URBs will be
| immediately completed, and this would result in the immediate
| period-elapsed calls right after the start, which may confuse
| applications.
|
| OTOH, submitting the data after silent URBs would, of course, result
| in a certain delay of the actual data processing, and this is rather
| more serious problem on modern systems, in practice.
|
| This patch tries to revert the workaround and lets the URB
| submission starting at PCM trigger for the playback again. As far
| as I've tested with various backends (native ALSA, PA, JACK, PW), I
| haven't seen any problems (famous last words :)
|
| Note that the capture stream handling needs no such workaround,
| since the capture is driven per received URB.
Link: https://lore.kernel.org/r/4e71531f-4535-fd46-040e-506a3c256bbd@marcan.st
Link: https://lore.kernel.org/r/s5hbl7li0fe.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210707112447.27485-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9ce650a75a.
This commit causes watchdog lockups on my machine, and while I have no
idea what the cause is, it bisected right to this commit, and reverting
the change promptly fixes it.
At least occasionally one of the watchdog call traces was
Call Trace:
_raw_spin_lock_irqsave+0x35/0x40
snd_pcm_period_elapsed+0x1b/0xa0 [snd_pcm]
snd_usb_endpoint_start+0x1a0/0x3c0 [snd_usb_audio]
start_endpoints+0x23/0x90 [snd_usb_audio]
snd_usb_substream_playback_trigger+0x7b/0x1a0 [snd_usb_audio]
snd_pcm_common_ioctl+0x1c44/0x2360 [snd_pcm]
snd_pcm_ioctl+0x2e/0x40 [snd_pcm]
__se_sys_ioctl+0x72/0xc0
do_syscall_64+0x4c/0xa0
entry_SYSCALL_64_after_hwframe+0x44/0xae
so presumably it's a locking error on that substream spinlock that
snd_pcm_period_elapsed() takes. But at this point I just want to have a
working system so that I can continue the merge window work tomorrow.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Clang warns:
sound/usb/mixer_scarlett_gen2.c:1189:32: warning: expression result
unused [-Wunused-value]
for (i = 0; i < count; i++, (u16 *)buf++)
^ ~~~~~
1 warning generated.
It appears the intention was to cast the void pointer to a u16 pointer
so that the data could be iterated through like an array of u16 values.
However, the cast happens after the increment because a cast is an
rvalue, whereas the post-increment operator only works on lvalues, so
the loop does not iterate as expected. This is not a bug in practice
because count is not greater than one at the moment but this could
change in the future so this should be fixed.
Replace the cast with a temporary variable of the proper type, which is
less error prone and fixes the iteration. Do the same thing for the
'u8 *' below this if block.
Fixes: ac34df733d ("ALSA: usb-audio: scarlett2: Update get_config to do endian conversion")
Link: https://github.com/ClangBuiltLinux/linux/issues/1408
Acked-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20210627051202.1888250-1-nathan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mixer control put callbacks should return 1 if the value is changed.
Fix the mute, air, phantom, direct monitor, speaker switch, talkback,
and MSD controls accordingly.
Fix scarlett2_speaker_switch_enable() to not ignore the return value
of scarlett2_sw_hw_change().
Reported-by: Aaron Wolf <aaron@wolftune.com>
Tested-by: Aaron Wolf <aaron@wolftune.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/76643f7ac81aef93351122d07881e30d51dcb1b9.1624798436.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 3 has 4 inputs with a pad control, not 2. Update
s18i8_gen3_info.pad_input_count.
Reported-by: Aaron Wolf <aaron@wolftune.com>
Tested-by: Aaron Wolf <aaron@wolftune.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/29a6ce412a42373daab7c96c395560461fcf08c6.1624798436.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For configuration items with a size of 16, scarlett2_usb_get_config()
was filling *buf with little-endian data. Update it to convert to CPU
endian. This function is not currently used so affects nothing yet;
will be used by the upcoming talkback feature.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/cbc8b6eedd859dd27086ab4126d724a86dd50bcb.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 and 18i20 Gen 3 support "speaker switching". Add a Speaker
Switch control which can be set to Off/Main/Alt.
When speaker switching is enabled or disabled, the interface may
change the state of the Analog Outputs 3 and 4 routing and the global
mute button, so use a flag private->speaker_switching_switched to note
that those should be checked when the next "monitor other"
notification is received.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/874193a534cd0aeb6f2e108ae761cadd2dc25ad2.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enabling/disabling speaker switching will update the mux
configuration. To prepare for this, add a private->mux_updated flag
and update the scarlett2_mux_src_enum_ctl_get() callback to check it.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/5ce3bb9fe4006b550d18c783c5ff640fe0bfbfcb.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Save the struct snd_kcontrol pointers for the sw_hw and mux controls.
This is in preparation for speaker switching support which needs to be
able to update those controls.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/269d89181bf29dbea80ba6f8cfff84fb23b77f86.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split part of scarlett2_sw_hw_enum_ctl_put() out into
scarlett2_sw_hw_change() so that the code which actually makes the
change is available in its own function. This will be used by the
speaker switching support which needs to set the SW/HW switch to HW
when speaker switching is enabled.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/f2cf91841ba067b490e7709bc4b14f4532b4ddd5.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Solo and 2i2 devices don't have a mixer but they do have a "direct
monitor" switch. Add support for getting and setting the state of this
switch.
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/61d23dc4feb3b046d870ad7203e66ff2bd1d278c.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some inputs on Gen 3 models support software-selectable phantom power.
Add support for getting and setting the state of those switches and
the "Phantom Power Persistence" switch.
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/5837ce8a8c686560fc8f40b4204dd2a10721869b.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some inputs on Gen 3 models have an "air" feature which can be enabled
from the driver or (model-dependent) from the front panel. Add support
for getting and setting the state of those switches.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/44d448a4150b9c068754759c9fdd2bfe21484487.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add initial support for the Focusrite Scarlett Solo and 2i2 devices:
- They have no mixer
- They don't support reporting sync status or levels
- The configuration space is laid out differently to the other models
- There is no level (line/inst) switch on input 1 of the Solo
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/190b90f6f1f8f8d4dfb5f0a7761ff8ae5c40fdde.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move scarlett2_usb_get() and scarlett2_usb_get_config() above the
functions relating to updating the configuration so that
scarlett2_usb_set_config() can call scarlett2_usb_get() in a
subsequent patch.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/1549f8e44548be679119f0b1462f888f4a03812d.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models allow the level and pad settings to be controlled from the
front-panel of the device. For these, the device will send an
"input-other" notification to prompt the driver to re-read the status
of those settings.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/06289a7697455e96b7dbdfd2d384d4b20f8df6e0.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current way of the scarlett2 mixer code managing the
usb_mixer_elem_info object is wrong in two ways: it passes its
internal index to the head.id field, and the val_type field is
uninitialized. This ended up with the wrong execution at the resume
because a bogus unit id is passed wrongly. Also, in the later code
extensions, we'll have more mixer elements, and passing the index will
overflow the unit id size (of 256).
This patch corrects those issues. It introduces a new value type,
USB_MIXER_BESPOKEN, which indicates a non-standard mixer element, and
use this type for all scarlett2 mixer elements, as well as
initializing the fixed unit id 0 for avoiding the overflow.
Tested-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/49721219f45b7e175e729b0d9d9c142fd8f4342a.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The scarlett2_ports struct contains both generic (hardware IDs and
descriptions) and model-specific (port count) data. Remove the generic
data from the scarlett2_device_info struct so it is not repeated for
every model.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/7a9e57e4e55a482390c692a9e60731d72b664a15.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Gen 3 devices do not put all of the mux entries for the same port
types together in order in the "set mux" message data. To prepare for
this, replace the struct scarlett2_ports num[] array and the
assignment_order[] array with mux_assignment[], a list of port types
and ranges that is defined in the struct scarlett2_device_info.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/08e8d784d78262cb57496d28ef1ad7b6213a90ab.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For each analogue output, in addition to the output volume (gain)
control, the hardware also has a mute control. Add ALSA mute controls
for each analogue output.
If the device has the line_out_hw_vol feature, then the mute control
is disabled along with the output volume control when the switch is
set to HW.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/6fad82174b44633e46cfd96332a038de74d544f2.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the USB device ID to the scarlett2_device_info struct so that the
switch statement which finds the appropriate struct can be replaced
with a loop that looks through an array of pointers to those structs.
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/474c408c29fb280a611e47e49e59ca2fb9810d27.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At extending the available mixer values for 32bit types, we forgot to
add the corresponding entries for the format dump in the proc output.
This may result in OOB access. Here adds the missing entries.
Fixes: bc18e31c30 ("ALSA: usb-audio: Fix parameter block size for UAC2 control requests")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210622090647.14021-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the hard-coded interface number and related constants for the
vendor-specific interface and look them up from the USB endpoint
descriptor.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164652.GA9237@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
scarlett2_usb_set_config() and scarlett2_usb_get_config() were copying
struct scarlett2_config. Use a pointer instead.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164648.GA9231@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix mixer control callbacks to use the correct members of the struct
snd_ctl_elem_value. The use of value.integer and value.enumerated were
swapped in a few places.
Update scarlett2_mux_src_enum_ctl_put() to use min() instead of
clamp() as value.enumerated.item is unsigned.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164647.GA9226@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mixer control put callbacks should return 1 if the value is changed.
Fix the sw_hw, level, pad, and button controls accordingly.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164645.GA9221@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The private->vol_updated flag was being checked outside of the
mutex_lock/unlock() of private->data_mutex leading to the volume data
being fetched twice from the device unnecessarily or old volume data
being returned.
Update scarlett2_*_ctl_get() and include the private->vol_updated flag
check inside the critical region.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164643.GA9216@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add index temporary variable to scarlett2_mixer_ctl_put() for
consistency with the other *_ctl_put() functions.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164641.GA9211@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename struct scarlett2_mixer_data to struct scarlett2_data. A
less-wordy name is better because it is used everywhere, and although
this is a mixer driver, it also controls other vendor-specific
features.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164639.GA9206@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To match the vendor's terminology, change #defines, identifiers, and
comments:
- mute/dim/hardware buttons are now called dim/mute
- mixer status/interrupt is now notify
- vol is now monitor
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164636.GA9199@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The per-model button_count value was used to determine whether
dim/mute controls should be added, but these are present iff
line_out_hw_vol is true. Remove button_count and replace with
SCARLETT2_BUTTON_MAX and a check for line_out_hw_vol true.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164634.GA9193@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just ignore instead of printing an error if the interrupt data is not
the expected length. This check was for development and the condition
has not been observed.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164632.GA9186@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 2 has 8 PCM Inputs, not 20. Fix the ports entry in
s18i8_gen2_info.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164625.GA9165@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 2 S/PDIF outputs are available at 192kHz, unlike
the 18i20 Gen 2. Remove the comment that says otherwise.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164622.GA9155@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This provides support for Denon DN-X1600 hardware mixer.
The device itself supports 44100, 48000 and 96000 (Hz)
sample rates, but switching rates via software is currently not working.
Therefore, this patch hardcodes the sample rate to 48000Hz which
enables all 8 channels to function correctly when the correct
sample rate is selected on the hardware itself.
MIDI also tested and works.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Tested-by: xalmoxis@gmail.com
Link: https://lore.kernel.org/r/20210610083528.603942-2-damien@zamaudio.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for retrieving the mux configuration from the hardware
when the driver is initialising. Previously the ALSA controls were
initialised to a default hard-coded state instead of being initialised
to match the hardware state.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Daniel Sales <daniel.sales.z@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/15b17c60a2bca174bcddcec41c9419b746f21c1d.1623091570.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for reading the mixer volumes from the hardware when the
driver is initialising. Previously these ALSA volume controls were
initialised to zero instead of being initialised to match the hardware
state.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Daniel Sales <daniel.sales.z@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/bb33fa9b79efc6f7a0f0e6fb7018cc8d4d59b3ba.1623091570.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver behaves a bit strangely for the playback stream --
namely, it starts sending silent packets at PCM prepare state while
the actual data is submitted at first when the trigger START is kicked
off. This is a workaround for the behavior where URBs are processed
too quickly at the beginning. That is, if we start submitting URBs at
trigger START, the first few URBs will be immediately completed, and
this would result in the immediate period-elapsed calls right after
the start, which may confuse applications.
OTOH, submitting the data after silent URBs would, of course, result
in a certain delay of the actual data processing, and this is rather
more serious problem on modern systems, in practice.
This patch tries to revert the workaround and lets the URB submission
starting at PCM trigger for the playback again. As far as I've tested
with various backends (native ALSA, PA, JACK, PW), I haven't seen any
problems (famous last words :)
Note that the capture stream handling needs no such workaround, since
the capture is driven per received URB.
Link: https://lore.kernel.org/r/20210601162457.4877-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM delay accounting in USB-audio driver is a bit complex to
follow, and this is an attempt to improve the readability and provide
some potential fix.
Basically, the PCM position delay is calculated from two factors: the
in-flight data on URBs and the USB frame counter. For the playback
stream, we advance the hwptr already at submitting URBs. Those
"in-flight" data amount is now tracked, and this is used as the base
value for the PCM delay correction. The in-flight data is decreased
again at URB completion in return. For the capture stream, OTOH,
there is no in-flight data, hence the delay base is zero.
The USB frame counter is used in addition for correcting the current
position. The reference frame counter is updated at each submission
and receiving time, and the difference from the current counter value
is taken into account.
In this patch, each in-flight data bytes is recorded in the new
snd_usb_ctx.queued field, and the total in-flight amount is tracked in
snd_usb_substream.inflight_bytes field, as the replacement of
last_delay field.
Note that updating the hwptr after URB completion doesn't work for
PulseAudio who tries to scratch the buffer on the fly; USB-audio is
basically a double-buffer implementation, hence the scratching the
buffer can't work for the already submitted data. So we always update
hwptr beforehand. It's not ideal, but the delay account should give
enough correctness.
Link: https://lore.kernel.org/r/20210601162457.4877-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a bunch of lines calculating the buffer size in bytes at
each time. Keep the value in subs->buffer_bytes and use it
consistently for the code simplicity.
Link: https://lore.kernel.org/r/20210601162457.4877-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power_state argument of snd_power_wait() is superfluous, receiving
only SNDRV_POWER_STATE_D0. Let's drop it in all callers for
simplicity.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210523090920.15345-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/mixer_scarlett_gen2.c:2000:5: warning: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static?
Fixes: 265d1a90e4 ("ALSA: usb-audio: scarlett2: Improve driver startup messages")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20210522180900.GA83915@f59a3af2f1d9
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.
Use info level instead of error for notifications.
Display the VID/PID so device_setup is targeted to the right device.
Display "enabled" message to easily confirm that the driver is loaded.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use usb_rcvctrlpipe() not usb_sndctrlpipe() for USB control input in
the Scarlett Gen 2 mixer driver. This fixes the device hang during
initialisation when used with the ehci-pci host driver.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/66a3d05dac325d5b53e4930578e143cef1f50dbe.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The direction of the pipe argument must match the request-type direction
bit or control requests may fail depending on the host-controller-driver
implementation.
Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe().
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: stable@vger.kernel.org # 5.10
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20210521133742.18098-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cur variable indicating the currently selected clock source can be
theoretically used as uninitialized after the recent commit
481f17c418 ("ALSA: usb-audio: Handle error for the current selector
gracefully"). For addressing it, initialize it before use.
Also, one place seems setting 0 to a wrong variable ret, instead of
cur; otherwise it makes little sense. Since the initialization is
done beforehand, we can get rid of this line, too.
Fixes: 481f17c418 ("ALSA: usb-audio: Handle error for the current selector gracefully")
Reported-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/4b261d68-f53f-240d-2d8a-2f88b337849d@canonical.com
Link: https://lore.kernel.org/r/s5hfsyhh97t.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The initialization of MIDI devices that are found on some LINE6
drivers are currently done in a racy way; namely, the MIDI buffer
instance is allocated and initialized in each private_init callback
while the communication with the interface is already started via
line6_init_cap_control() call before that point. This may lead to
Oops in line6_data_received() when a spurious event is received, as
reported by syzkaller.
This patch moves the MIDI initialization to line6_init_cap_control()
as well instead of the too-lately-called private_init for avoiding the
race. Also this reduces slightly more lines, so it's a win-win
change.
Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com
Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com
Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo
Cc: Hyeonggon Yoo <42.hyeyoo@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210518083939.1927-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we bail out when the device returns an error or an invalid
value for the current clock selector value via
uac_clock_selector_get_val(). But it's possible that the device is
really uninitialized and waits for the setup of the proper route at
first.
For handling such a case, this patch lets the driver dealing with the
error or the invalid error more gracefully, choosing the clock source
automatically instead.
Link: https://lore.kernel.org/r/20210518152112.8016-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just does refactoring of the UAC2/3 clock setup code.
There should be no functional changes. The major changes are:
* Provide union objects for pointing both UAC2 and UAC3 objects
* Unify clock source, selector and multiplier helper functions
* Unify __uac_clock_find_source() to deal with both UAC2 and UAC3
equally
Link: https://lore.kernel.org/r/20210518152112.8016-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor code refactoring by merging the superfluous function calls.
The functions were split in the past for covering pre-history USB
driver code, but this is utterly useless.
Link: https://lore.kernel.org/r/20210517131545.27252-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently us428ctls_shmem pages are allocated dynamically upon the
mmap call, but this is quite racy. Since the shared memory itself is
mandatory for the mmap, let's allocate it at the beginning of the card
initialization. Also, fix the initialization of the wait queue, too.
Link: https://lore.kernel.org/r/20210517131545.27252-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM shmem pages are allocated in snd_usx2y_usbpcm_prepare().
Theoretically the prepare callback may be called simultaneously for
both playback and capture, hence this allocation can be racy.
Make sure that the allocation is performed exclusively by extending
the pcm_mutex lock to cover the allocation code, too.
Link: https://lore.kernel.org/r/20210517131545.27252-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Theoretically the initialization functions in usx2y drivers may be
called multiple times as the driver gets initialized via hwpdep
ioctl. Meanwhile, those functions including memory allocations don't
check whether they are called twice, and they forget the old
resources, which would lead to memory leaks.
This patch adds the sanity checks about the doubly initializations to
give kernel WARNING, and returns an error in such a case. Also, each
allocation assures to release the resources at its error path
properly.
Link: https://lore.kernel.org/r/20210517131545.27252-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The initialization os usx2y driver is multi-staged, and the PCM and
other device creations are done after the DSP is loaded and
initialized. Upon the initialization, when an error happens, the
driver tries to call snd_card_free(). But this is dangerous, and in
general, the driver cannot kill itself during its operation.
Hence better to drop the snd_card_free() call from there.
Link: https://lore.kernel.org/r/20210517131545.27252-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usx2y drivers may expose the allocated pages via mmap, but it performs
zero-clear only for the struct size, not aligned with the page size.
This leaves out some uninitialized trailing bytes.
This patch fixes the clearance to cover all memory that are exposed to
user-space.
Link: https://lore.kernel.org/r/20210517131545.27252-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes various trivial coding-style issues in usx2y code,
such as:
* the assginments in if condition
* comparison order with constants
* NULL / zero checks
* unsigned -> unsigned int
* addition of braces in control blocks
* debug print with function names
* move local variables in block into function head
* reduction of too nested indentations
No functional changes.
Link: https://lore.kernel.org/r/20210517131545.27252-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch corrects merely the spaces in the usx2y code, including the
superfluous trailing space in the debug prints and a slight reformat
of some comment lines. Nothing really touches about the code itself.
Link: https://lore.kernel.org/r/20210517131545.27252-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For improving readability, convert camelCase fields, variables and
functions to the plain names with underscore. Also align the macros
to be capital letters.
All done via sed, no functional changes.
Note that you'll still see many coding style issues even after this
patch; the fixes will follow.
Link: https://lore.kernel.org/r/20210517131545.27252-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced MIDI endpoint parser code has an access to the
field without the size validation, hence it might lead to
out-of-bounce access. Add the sanity checks for the descriptor
sizes.
Fixes: eb596e0fd1 ("ALSA: usb-audio: generate midi streaming substream names from jack names")
Link: https://lore.kernel.org/r/20210511090500.2637-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable len is set to zero but this value is never read as it is
overwritten with a new value later on, hence it is a redundant
assignment and can be removed.
Cleans up the following clang-analyzer warning:
sound/usb/mixer.c:2713:3: warning: Value stored to 'len' is never read
[clang-analyzer-deadcode.DeadStores].
Reported-by: Abaci Robot <abaci@linux.alibaba.com>
Signed-off-by: Jiapeng Chong <jiapeng.chong@linux.alibaba.com>
Link: https://lore.kernel.org/r/1619519194-57806-1-git-send-email-jiapeng.chong@linux.alibaba.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent endpoint management change for implicit feedback mode added
a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at
snd_usb_endpoint_stop() to assure no leftover for the feedback from
the already stopped capture stream. This turned out to cause a
regression, however, when full-duplex streams were running and only a
capture was stopped. Because of the above clearance of ep->sync_sink
pointer, no more feedback is done, hence the playback will stall.
This patch fixes the ep->sync_sink clearance to be done only after all
endpoints are released, for addressing the regression.
Reported-and-tested-by: Lucas Endres <jaffa225man@gmail.com>
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426063349.18601-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the examinations and experiments with lots of Roland and BOSS
USB-audio devices, we found out that the recently introduced
full-duplex operations with the implicit feedback mode work fine for
quite a few devices, while the others need only the capture-side quirk
to enforce the full-duplex mode. The recent commit d86f43b17e
("ALSA: usb-audio: Add support for many Roland devices' implicit
feedback quirks") tried to add such quirk entries manually in the
lists, but this turned out to be too many and error-prone, hence it
was reverted again.
This patch is another attempt to cover those missing Roland/BOSS
devices but in a more generic way. It matches the devices with the
vendor ID 0x0582, and checks whether they are with both ASYNC sync
types or ASYNC is only for capture device. In the former case, it's
the device with the implicit feedback mode, and applies accordingly.
In both cases, the capture stream requires always the full-duplex
mode, and we apply the known capture quirk for that, too.
Basically the already existing BOSS device quirk entries become
redundant after this generic matching, so those are removed. Although
the capture_implicit_fb_quirks[] table became empty and superfluous, I
keep it for now, so that people can put a special device easily at any
time later again.
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519
Tested-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/20210422120413.457-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit d86f43b17e ("ALSA: usb-audio: Add support for
many Roland devices' feedback quirks").
It turned out that many quirk entries there don't contain the proper
EP values and/or the quirk types, which lead to the broken
operations.
As we're going to cover all Roland/BOSS devices in a more generic way
rather the explicit lists, let's revert the previous additions at
first.
Fixes: d86f43b17e ("ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks")
Link: https://lore.kernel.org/r/20210422120413.457-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL)
the error return path returns -ENOMEM via the exit label "error". Other
uses of the same error exit label set the err variable to -ENOMEM but this
is not being used. I believe the original intent was for the error exit
path to return the value in err rather than the hard coded -ENOMEM, so
return this rather than the hard coded -ENOMEM.
Addresses-Coverity: ("Unused value")
Fixes: 738d9edcfd ("ALSA: usb-audio: Add sanity checks for invalid EPs")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210420134719.381409-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer devices are supposed to be working with the implicit feedback
mode, but so far the attempt to apply the implicit feedback caused
issues, hence we explicitly skipped the implicit feedback mode for
them. Recently, Geraldo discovered that the device actually works if
you skip the generic matching of the sync EPs for the capture stream.
That is, we should apply the implicit feedback setup for the playback
like other similar devices, while we need to return 1 from
audioformat_capture_quirk() so that no further matching will be done.
And, later on, Olivia reported later that the fiddling with the
capture quirk alone doesn't suffice for the test with speaker-test
program. This seems to be a similar case like the recently fixed BOSS
devices. Indeed, the problem could be addressed by setting
playback_first flag, which indicates that the playback URBs have to be
sent out at first even in the implicit feedback mode.
This patch implements the application of the implicit feedback to
Pioneer devices as described in the above. The former
skip_pioneer_sync_ep() was dropped, and instead we provide
is_pioneer_implicit_fb() to check the Pioneer devices that need the
implicit feedback. In the audioformat_implicit_fb_quirk(), simply
apply the implicit fb for playback and set chip->playback_first flag
if matching, and in audioformat_capture_quirk()(), it returns 1 for
skipping the generic EP sync handling.
Reported-by: Geraldo <geraldogabriel@gmail.com>
Tested-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de
Link: https://lore.kernel.org/r/20210419153918.450-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add case statement to set sample-rate for the DJM-750 Pioneer
mixer. This was included as part of another patch but I think it has
been archived on Patchwork and hasn't been merged.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210418165901.25776-1-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes USB audio capture and playback possible and pristine on my Roland
INTEGRA-7, Boutique D-05, and R-26, along with many more I've encountered
people having had issues with over the last decade or so.
Signed-off-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the recent rewrite of the implicit feedback support, we've
tested to apply the implicit fb on BOSS devices, but it failed, as the
capture stream didn't start without the playback. As the end result,
it got another type of quirk for tying both streams but starts
playback always (commit 6234fdc1ce "ALSA: usb-audio: Quirk for BOSS
GT-001").
Meanwhile, Mike Oliphant has tested the real implicit feedback mode
for the playback again with the latest code, and found out that it
actually works if the initial feedback sync is skipped; that is, on
those BOSS devices, the playback stream has to be started at first
without waiting for the capture URB completions. Otherwise it gets
stuck. In the rest operations after the capture stream processed, we
can take them as the implicit feedback source.
This patch is an attempt to improve the support for BOSS devices with
the implicit feedback mode in the way described above. It adds a new
flag to snd_usb_audio, playback_first, indicating that the playback
stream starts without sync with the initial capture completion. This
flag is set in the quirk table with the new IMPLICIT_FB_BOTH type.
Reported-and-tested-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, we have some assumption that the audio clock
selector has been set up implicitly and don't want to touch it unless
it's really needed for the fallback autoclock setup. This works for
most devices but some seem having a problem. Partially this was
covered for the devices with a single connector at the initialization
phase (commit 086b957cc1 "ALSA: usb-audio: Skip the clock selector
inquiry for single connections"), but also there are cases where the
wrong clock set up is kept silently. The latter seems to be the cause
of the noises on Behringer devices.
In this patch, we explicitly set up the audio clock selector whenever
the appropriate node is found.
Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327
Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UA-101 device and co are supported by another driver, snd-ua101, but
the USB audio class driver (snd-usb-audio) catches all and this
resulted in the lack of functionality like missing MIDI devices.
This patch introduces a sort of deny-listing for those devices to just
return -ENODEV at probe in snd-usb-audio driver, so that it falls back
to the probe by snd-ua101.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212477
Link: https://lore.kernel.org/r/20210408075656.30184-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few calls of usb_driver_claim_interface() but all of those
miss the proper error checks, as reported by Coverity. This patch
adds those missing checks.
Along with it, replace the magic pointer with -1 with a constant
USB_AUDIO_IFACE_UNUSED for better readability.
Reported-by: coverity-bot <keescook+coverity-bot@chromium.org>
Addresses-Coverity-ID: 1475943 ("Error handling issues")
Addresses-Coverity-ID: 1475944 ("Error handling issues")
Addresses-Coverity-ID: 1475945 ("Error handling issues")
Fixes: b1ce7ba619 ("ALSA: usb-audio: claim autodetected PCM interfaces all at once")
Fixes: e5779998bf ("ALSA: usb-audio: refactor code")
Link: https://lore.kernel.org/r/202104051059.FB7F3016@keescook
Link: https://lore.kernel.org/r/20210406113534.30455-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'tags/mute-led-rework' into for-next
ALSA: control - add generic LED API
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Logitech ConferenceCam Connect is a compound USB device with UVC and
UAC. Not 100% reproducible but sometimes it keeps responding STALL to
every control transfer once it receives get_freq request.
This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419
Signed-off-by: Ikjoon Jang <ikjn@chromium.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear Mic on Lenovo P620 cannot record after S3, despite that there's no
error and the other two functions of the USB audio, Line In and Line
Out, work just fine.
The mic starts to work again after running userspace app like "alsactl
store". Following the lead, the evidence shows that as soon as connector
status is queried, the mic can work again.
So also check connector value on resume to "wake up" the USB audio to
make it functional.
This can be device specific, however I think this generic approach may
benefit more than one device.
Now the resume callback checks connector, and a new callback,
reset_resume, to also restore switches and volumes.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210325165918.22593-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Majority of changes are various ASoC device/platform-specific small
fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are usual HD-audio quirks.
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Merge tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The majority of changes are various ASoC device/platform-specific
small fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are the usual HD-audio quirks"
* tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (44 commits)
ALSA: usb-audio: Fix unintentional sign extension issue
ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
ASoC: rt711: add snd_soc_component remove callback
ASoC: rt5659: Update MCLK rate in set_sysclk()
ASoC: simple-card-utils: Do not handle device clock
ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8
ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8
ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro
ALSA: hda/realtek: Apply headset-mic quirks for Xiaomi Redmibook Air
ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
ALSA: dice: fix null pointer dereference when node is disconnected
ALSA: hda: generic: Fix the micmute led init state
ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
spi: cadence: set cqspi to the driver_data field of struct device
ASoC: SOF: intel: fix wrong poll bits in dsp power down
ASoC: codecs: wcd934x: add a sanity check in set channel map
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: remove remnants of sirf prima/atlas audio codec
...
The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.
Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c79 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MODULE_SUPPORTED_DEVICE was added in pre-git era and never was
implemented. We can safely remove it, because the kernel has grown
to have many more reliable mechanisms to determine if device is
supported or not.
Signed-off-by: Leon Romanovsky <leonro@nvidia.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The problem was in wrong "if" placement. chip->quirk_type is freed
in snd_card_free_when_closed(), but inside if statement it's accesed.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/16da19126ff461e5e64a9aec648cce28fb8ed73e.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear audio on Lenovo ThinkStation P620 stops working after commit
1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo
ThinkStation P620"):
[ 6.013526] usbcore: registered new interface driver snd-usb-audio
[ 6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
I overlooked the issue because when I was working on the said commit,
only the front audio is tested. Apology for that.
Changing supports_autosuspend in driver is too late for disabling
autosuspend, because it was already used by USB probe routine, so it can
break the balance on the following code that depends on
supports_autosuspend.
Fix it by using usb_disable_autosuspend() helper, and balance the
suspend count in disconnect callback.
Fixes: 1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304043419.287191-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai