Variable length array is used in 'snd_us16x08_meter_get()', while there
is no need. It's better to purge it because variable length array has
overhead for stack handling.
This commit replaces the array with static length. Sparse generated below
warning.
sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used.
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new structures as const as they are only passed as
an argument to the function add_new_ctl. This agrument is of type const,
so snd_kcontrol_new structures having this property can be made const
too.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed inner a file, functions should have static qualifier for
local-linkage.
This commit fixes the bug. Sparse generated below warning.
sound/usb/mixer_us16x08.c:1043:32: warning: symbol 'snd_us16x08_create_meter_store' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed by one referrer inner a file, variables should have static
qualifier to declare local-linkage.
This commit fixes the bug. Sparse generated below warnings.
sound/usb/mixer_us16x08.c:156:13: warning: duplicate const
sound/usb/mixer_us16x08.c:156:18: warning: symbol 'route_names' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for Tascam US-16x08 usb interface.
Even that this is an usb compliant device,
the input channels and DSP functions (EQ/Compressor) aren't accessible
by default.
Signed-off-by: Detlef Urban <onkel@paraair.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While not all line6 devices currently support PCM, it causes no
harm to 'have it prepared'.
This also fixes toneport, which only has PCM - in which case
we previously skipped the USB transfer properties detection completely.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit f6a0dd107a.
The commit caused a regression on LINE6 Transport that has no control
caps. Although reverting the commit may result back in a spurious
error message for some device again, it's the simplest regression fix,
hence it's taken as is at first. The further code fix will follow
later.
Fixes: f6a0dd107a ("ALSA: line6: Only determine control port properties if needed")
Reported-by: Igor Zinovev <zinigor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for both little endian (DSD_U32_LE) and big endian
(DSD_U32_BE) version of the Amanero firmware.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 16200948d8.
The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.
As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.
Fixes: 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.
Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).
- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track
[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.
[Solution]
This patch adds DSD silence pattern to the endpoint settings.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the following devices.
- TEAC NT-503
- TEAC UD-503
- TEAC UD-501
(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
DSD mode, same as Denon/Marantz devices.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the
following warning.
[ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072),
cval->res is probably wrong.
[ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val =
4608/7680/1
Adding it to the list of devices in volume_control_quirks makes it work
properly, fixing related typo.
Signed-off-by: Con Kolivas <kernel@kolivas.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
since commit 57e6dae108 ("ALSA: usb-audio: do not trust too-big
wMaxPacketSize values"), the expected packetsize is always limited
to nominal + 25%. It was discovered, that some devices (Android audio
accessory) have a much higher jitter in used packetsizes than 25%
which would result in BABBLE condition and dropping of packets.
A better solution is so assume the jitter to be the nominal packetsize:
-one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape <apape@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a kernel crash report showing like:
Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000
[00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000
Internal error: Oops: 17 [#1] PREEMPT SMP ARM
CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1
task: a3ae61c0 ti: a08c8000 task.ti: a08c8000
PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio]
LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio]
pc : [<7f0eb22c>] lr : [<7f0e57fc>] psr: 200e0193
sp : a08c9c98 ip : a08c9ce8 fp : a08c9ce4
r10: 0000000a r9 : 00000102 r8 : 94cb3000
r7 : 94cb3000 r6 : 94d0f000 r5 : 94d0e8e8 r4 : 94d0e000
r3 : 7f0eb21c r2 : 00000000 r1 : 94cb3000 r0 : 00000000
Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user
Control: 10c5387d Table: 31d7c04a DAC: 00000015
Process dbus-daemon (pid: 250, stack limit = 0xa08c8238)
Stack: (0xa08c9c98 to 0xa08ca000)
...
Backtrace:
[<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio])
[<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4)
[<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0)
[<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148)
[<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380)
[<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc)
[<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8)
[<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8)
[<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78)
[<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100)
[<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8)
[<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c)
[<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0)
[<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20)
Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008)
Kernel panic - not syncing: Fatal exception in interrupt
There is a race between retire_capture_urb() and stop_endpoints().
The latter is called at stopping the stream and it sets some endpoint
fields to NULL. But its call is asynchronous, thus the pending
complete callback might get called after these NULL clears, and it
leads the NULL dereference like the above.
The fix is to move the NULL clearance after the synchronization,
i.e. wait_clear_urbs(). This is called at prepare and hw_free
callbacks, so it's assured to be called before the restart of the
stream or the release of the stream.
Also, while we're at it, put the EP_FLAG_RUNNING flag check at the
beginning of snd_complete_urb() to skip the pending complete after the
stream is stopped.
Fixes: b2eb950de2 ("ALSA: usb-audio: stop both data and sync...")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Reported-by: Mark Craske <Mark_Craske@mentor.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function returns -EINVAL even if it builds the stream properly.
The bogus error code sneaked in during the code refactoring, but it
wasn't noticed until now since the returned error code itself is
ignored in anyway. Kill it here, but there is no behavior change by
this patch, obviously.
Fixes: e5779998bf ('ALSA: usb-audio: refactor code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Userspace apps have to claim USB interfaces before using endpoints in
them (drivers/usb/core/devio.c:checkintf()). It's a lock mechanism so
that two "drivers" don't steal data from each other. Kernel drivers don't
have to claim interfaces to work - but they should, to lock out userspace.
While there, fix line6_properties struct to match checkpatch.pl.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver implements the deferred device disconnection for
the device in use. In this mode, the disconnection callback returns
immediately while the actual ALSA card object removal happens later
when all files get closed. As Shuah reported, this code flow,
however, leads to a use-after-free, detected by KASAN:
BUG: KASAN: use-after-free in snd_usb_audio_free+0x134/0x160 [snd_usb_audio] at addr ffff8801c863ce10
Write of size 8 by task pulseaudio/2244
Call Trace:
[<ffffffff81b31473>] dump_stack+0x67/0x94
[<ffffffff81564ef1>] kasan_object_err+0x21/0x70
[<ffffffff8156518a>] kasan_report_error+0x1fa/0x4e0
[<ffffffff81564ad7>] ? kasan_slab_free+0x87/0xb0
[<ffffffff81565733>] __asan_report_store8_noabort+0x43/0x50
[<ffffffffa0fc0f54>] ? snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0f54>] snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0fb1>] snd_usb_audio_dev_free+0x31/0x40 [snd_usb_audio]
[<ffffffff8243c78a>] __snd_device_free+0x12a/0x210
[<ffffffff8243d1f5>] snd_device_free_all+0x85/0xd0
[<ffffffff8242cae4>] release_card_device+0x34/0x130
[<ffffffff81ef1846>] device_release+0x76/0x1e0
[<ffffffff81b37ad7>] kobject_release+0x107/0x370
.....
Object at ffff8801c863cc80, in cache kmalloc-2048 size: 2048
Allocated:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff8156450d>] kasan_kmalloc+0xad/0xe0
[<ffffffff81560d1a>] kmem_cache_alloc_trace+0xfa/0x240
[<ffffffff8214ea47>] usb_alloc_dev+0x57/0xc90
[<ffffffff8216349d>] hub_event+0xf1d/0x35f0
....
Freed:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff81564ac1>] kasan_slab_free+0x71/0xb0
[<ffffffff81560929>] kfree+0xd9/0x280
[<ffffffff8214de6e>] usb_release_dev+0xde/0x110
[<ffffffff81ef1846>] device_release+0x76/0x1e0
....
It's the code trying to clear drvdata of the assigned usb_device where
the usb_device itself was already released in usb_release_dev() after
the disconnect callback.
This patch fixes it by checking whether the code path is via the
disconnect callback, i.e. chip->shutdown flag is set.
Fixes: 79289e2419 ('ALSA: usb-audio: Refer to chip->usb_id for quirks...')
Reported-and-tested-by: Shuah Khan <shuahkh@osg.samsung.com>
Cc: <stable@vger.kernel.org> # v4.6+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error checking here is messed up so we could end up dereferencing
-EFAULT.
Fixes: a16039cbf1 ('ALSA: line6: Add hwdep interface to access the POD control messages')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit c039aaa77a was incomplete,
missing part of the setup for Live. This makes also audio input work,
in addition to audio output.
Fixes: c039aaa77a
Reported-by: Eddi De Pieri <eddi@depieri.net>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, usb-line6 module exports an array of MIDI manufacturer ID and
usb-pod module uses it. However, the declaration is not the definition in
common header. The difference is explicit length of array. Although
compiler calculates it and everything goes well, it's better to use the
same representation between definition and declaration.
This commit fills the length of array for usb-line6 module. As a small
good sub-effect, this commit suppress below warnings from static analysis
by sparse v0.5.0.
sound/usb/line6/driver.c:274:43: error: cannot size expression
sound/usb/line6/driver.c:275:16: error: cannot size expression
sound/usb/line6/driver.c:276:16: error: cannot size expression
sound/usb/line6/driver.c:277:16: error: cannot size expression
Fixes: 705ececd1c ("Staging: add line6 usb driver")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DragonFly quirk added in 42e3121d90 ("ALSA: usb-audio: Add a more
accurate volume quirk for AudioQuest DragonFly") applies a custom dB map
on the volume control when its range is reported as 0..50 (0 .. 0.2dB).
However, there exists at least one other variant (hw v1.0c, as opposed
to the tested v1.2) which reports a different non-sensical volume range
(0..53) and the custom map is therefore not applied for that device.
This results in all of the volume change appearing close to 100% on
mixer UIs that utilize the dB TLV information.
Add a fallback case where no dB TLV is reported at all if the control
range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also
restrict the quirk to only apply to the volume control as there is also
a mute control which would match the check otherwise.
Fixes: 42e3121d90 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly")
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: David W <regulars@d-dub.org.uk>
Tested-by: David W <regulars@d-dub.org.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.
NULL check before some freeing functions is not needed.
Based on checkpatch warning
"kfree(NULL) is safe this check is probably not required"
and kfreeaddr.cocci by Julia Lawall.
Generated by: scripts/coccinelle/free/ifnullfree.cocci
CC: Andrej Krutak <dev@andree.sk>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We must do it this way, because e.g. POD X3 won't play any sound unless
the host listens on the bulk EP, so we cannot export it only via libusb.
The driver currently doesn't use the bulk EP messages in other way,
in future it could e.g. sense/modify volume(s).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only initialize PCM for POD HD devices that support it.
No POD HD seems to support MIDI, thus drop the initialization.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes audio in/out and basic initialization via control EP (emulates
what original driver does). The initialization is done similarly to original
POD, firmware and serial IDs are read and exported via sysfs.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all PODs use MIDI via USB data interface, thus allow avoiding
that code and instead using direct processing.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
POD X3 can initialize similarly to older PODs, but it doesn't have the MIDI
interface. Instead, configuration is done via proprietary bulk EP messages.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. POD X3 seems to require playback data to be sent to it to generate
capture data. Otherwise the device stalls and doesn't send any more capture
data until it's reset.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Splits max_packet_size to max_packet_size_in/out (e.g. for
different channel counts).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This has two parts:
* intervals_per_second setup
(high speed needs 8000, instead of 1000)
* iso_buffers setup (count of iso buffers depends on
USB speed, 2 is not enough for high speed)
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This basically changes LINE6_ISO_BUFFERS constant to a configurable
iso_buffers property.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on B850V3 CP2114 as it is unsupported and
causes noisy "current rate is different from the runtime rate" messages
when playback starts.
Signed-off-by: Ken Lin <ken.lin@advantech.com.tw>
Signed-off-by: Akshay Bhat <akshay.bhat@timesys.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc already print similar error once failing to alloc
enough memory, so let's remove this dump here.
Signed-off-by: Shawn Lin <shawn.lin@rock-chips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Users of devices affected by the Tenor feedback data error report
buffer underruns, even with the +/- 0x1.0000 quirk applied.
Compensating the error with 0xf000 instead seems to reliably fix
that issue.
See
https://sourceforge.net/p/alsa/mailman/message/35230259/
Reported-and-tested-by: Norman Nolte <norman.nolte@gmx.net>
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to
more that are based on the Tenor 8802TL chipset. Devices built by T+A
are affected too, and they apparently all use the same USB PID:PID.
Extend the quirky handling for that device as well, and rename the
quirks flag.
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
That's a quirk, after all, so move it where to all the other quirks
live.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 02fc76f6a changed base of the sysfs attributes from device to card.
The "show" callbacks dereferenced wrong objects because of this.
Fixes: 02fc76f6a7 ('ALSA: line6: Create sysfs via snd_card_add_dev_attr()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's an error, pcm is released in line6_pcm_acquire already.
Fixes: 247d95ee6d ('ALSA: line6: Handle error from line6_pcm_acquire()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc will print enough information in case of failure.
Signed-off-by: Wolfram Sang <wsa-dev@sang-engineering.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ELP HD USB Camera (05a3:9420) needs this quirk for suppressing
the unsupported sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VF0610 does not support reading the sample rate which leads to many
lines of "cannot get freq at ep 0x82". This patch adds the USB ID
(0x041E:4080) to snd_usb_get_sample_rate_quirk() list.
Signed-off-by: Piotr Karasinski <peter.karasinski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To allow for structure randomisation, replace the in order struct
initialisation style with explicit field style.
The Coccinelle semantic patch used to make this change is as follows:
@decl@
identifier i1,fld;
type T;
field list[n] fs;
@@
struct i1 {
fs
T fld;
...};
@@
identifier decl.i1,i2,decl.fld;
expression e;
position bad.p, bad.fix;
@@
struct i1 i2@p = { ...,
+ .fld = e
- e@fix
,...};
Signed-off-by: Amitoj Kaur Chawla <amitoj1606@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many USB audio devices with buggy firmware that don't react
with the sample rate reading properly. This often results in the
flood of error messages and slowing down the operation.
The sample rate read back is basically only for confirming the sample
rate setup, and it's not critically important. As a compromise, in
this patch, we stop the sample rate read back once when the device
gives errors more than tolerance (twice, as of now). This should
improve most of error cases while we still can catch the firmware
bugginess.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
we've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy.
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Merge tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"We've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy"
* tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix inconsistent monitor_present state until repoll
ALSA: hda - Fix regression of monitor_present flag in eld proc file
ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
ALSA: sscape: Use correct format identifier for size_t
ALSA: usb-audio: Add a quirk for Plantronics BT300
ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
ALSA: hda - Bind with i915 only when Intel graphics is present
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.
[Some background: the patch essentially skips the controls of a couple
of FU volumes. Although the firmware exposes the dB and the value
information via the usb descriptor, changing the values (we set the
min volume as default) screws up the device. Although this has been
fixed in the newer firmware, the devices are shipped with the old
firmware, thus we need the workaround in the driver side. -- tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
miniDSP USBStreamer UAC2 devices send clock validity changes with the
control field set to zero. The current interrupt handler ignores all
packets if the control field does not match the mixer element's, but
it really should only do that in case that field is needed to
distinguish multiple elements with the same ID.
This patch implements a logic that lets notifications packets pass
if the element ID is unique for a given device.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 specifies clock sources that optionally have validity controls.
This patch exposes them as mixer controls, so they can be read (and
at least in theory even be written) by userspace applications in order
to make clock selection policy decisions.
This implementation does nothing if the device is not UAC2 compliant,
or if the clock source does not define said validity control bits.
Tested with a miniDSP USBStreamer (0x2752/0x0016).
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:
"Some bug fixes on au0828 and snd-usb-audio:
- the au0828+snd-usb-audio MC patch broke several things and produced
some race conditions. Better to revert the patches, and re-work on
them for a next version
- fix a regression at tuner disable links logic
- properly handle dev_state as a bitmask"
* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] Revert "[media] media: au0828 change to use Managed Media Controller API"
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
[media] au0828: Fix dev_state handling
[media] au0828: fix au0828_v4l2_close() dev_state race condition
[media] media: au0828 fix to clear enable/disable/change source handlers
[media] v4l2-mc: cleanup a warning
[media] au0828: disable tuner links and cache tuner/decoder
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
usb 2-1.3: 2:1: cannot get freq at ep 0x85
usb 2-1.3: 1:1: cannot get freq at ep 0x2
....
Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes:
- A fix in ALSA timer core to avoid possible BUG() trigger
- A fix in ALSA timer core 32bit compat layer
- A few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- Fixes of USB-audio double-free at some error paths
- A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- a fix in ALSA timer core to avoid possible BUG() trigger
- a fix in ALSA timer core 32bit compat layer
- a few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- fixes of USB-audio double-free at some error paths
- a fix for memory leak in DICE driver at hotunplug"
* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: timer: Use mod_timer() for rearming the system timer
ALSA: hda - fix front mic problem for a HP desktop
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
ALSA: dice: fix memory leak when unplugging
ALSA: hda - Apply fix for white noise on Asus N550JV, too
ALSA: hda - Fix white noise on Asus N750JV headphone
ALSA: hda - Asus N750JV external subwoofer fixup
ALSA: timer: fix gparams ioctl compatibility for different architectures
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
https://bugzilla.kernel.org/show_bug.cgi?id=115561
It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.
So, better to revert it and fix the core before reapplying this
change.
This reverts commit aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed.
So this request came up: a collection of small fixes to paper over
the inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too.
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Merge tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed. So
this request came up: a collection of small fixes to paper over the
inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too"
* tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix missing ELD update at unplugging
ALSA: usb-audio: add Microsoft HD-5001 to quirks
ALSA: hda - Workaround for unbalanced i915 power refcount by concurrent probe
ALSA: hda - Fix spurious kernel WARNING on Baytrail HDMI
ALSA: hda - Fix forgotten HDMI monitor_present update
ALSA: hda - Really restrict i915 notifier to HSW+
The Microsoft HD-5001 webcam microphone does not support sample rate
reading as the HD-5000 one.
This results in dmesg errors and sound hanging with pulseaudio.
Signed-off-by: Victor Clément <victor.clement@openmailbox.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514
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Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"
* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
ALSA: mixart: silence an uninitialized variable warning
ALSA: usb-audio: Add sanity checks for endpoint accesses
ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
ALSA: hda - Limit i915 HDMI binding only for HSW and later
ALSA: hda - Fix unconditional GPIO toggle via automute
ALSA: mixart: silence unitialized variable warnings
ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
ASoC: rsnd: add simplified module explanation
ASoC: hdac_hdmi: Add broxton device ID
ASoC: Intel: Bxtn: Add Broxton PCI ID
ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
ASoC: Intel: add dmabuffer to common sst_dsp
ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
ASoC: Intel: Skylake: Fix whitepsace issues
ASoC: Intel: Skylake: Move module id defines
...
Add some sanity check codes before actually accessing the endpoint via
get_endpoint() in order to avoid the invalid access through a
malformed USB descriptor. Mostly just checking bNumEndpoints, but in
one place (snd_microii_spdif_default_get()), the validity of iface and
altsetting index is checked as well.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create_fixed_stream_quirk() may cause a NULL-pointer dereference by
accessing the non-existing endpoint when a USB device with a malformed
USB descriptor is used.
This patch avoids it simply by adding a sanity check of bNumEndpoints
before the accesses.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* commit '840f5b0572ea': (381 commits)
media: au0828 disable tuner to demod link in au0828_media_device_register()
[media] touptek: cast char types on %x printk
[media] touptek: don't DMA at the stack
[media] mceusb: use %*ph for small buffer dumps
[media] v4l: exynos4-is: Drop unneeded check when setting up fimc-lite links
[media] v4l: vsp1: Check if an entity is a subdev with the right function
[media] hide unused functions for !MEDIA_CONTROLLER
[media] em28xx: fix Terratec Grabby AC97 codec detection
[media] media: add prefixes to interface types
[media] media: rc: nuvoton: switch attribute wakeup_data to text
[media] v4l2-ioctl: fix YUV422P pixel format description
[media] media: fix null pointer dereference in v4l_vb2q_enable_media_source()
[media] v4l2-mc.h: fix yet more compiler errors
[media] staging/media: add missing TODO files
[media] media.h: always start with 1 for the audio entities
[media] sound/usb: Use meaninful names for goto labels
[media] v4l2-mc.h: fix compiler warnings
[media] media: au0828 audio mixer isn't connected to decoder
[media] sound/usb: Use Media Controller API to share media resources
[media] dw2102: add support for TeVii S662
...
Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. Media Controller
specific initialization is done after sound card is registered. ALSA
creates Media interface and entity function graph nodes for Control,
Mixer, PCM Playback, and PCM Capture devices.
snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is
granted, it will release it from snd_usb_hw_free(). If resource is
busy, -EBUSY is returned.
Media specific cleanup is done in usb_audio_disconnect().
Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
Plantronics DA45 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x4" and "cannot get freq at
ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'umidi' object will be free'd on the error path by snd_usbmidi_free()
when tearing down the rawmidi interface. So we shouldn't try to free it
in snd_usbmidi_create() after having registered the rawmidi interface.
Found by KASAN.
Signed-off-by: Andrey Konovalov <andreyknvl@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam HD-6000 (045e:076f) requires the similar quirk for
avoiding the stall due to the invalid sample rate reads.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=111491
Signed-off-by: Lev Lybin <lev.lybin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got
through. This patch fixes the vendor ID and aligns the comment.
Fixes: a4eae3a506 ('ALSA: usb: Add native DSD support for Oppo HA-1')
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new option "quirk_alias" to snd-usb-audio driver for
allowing user to pass the quirk alias list. A quirk alias consists of
a string form like 0123abcd:5678beef, which makes to apply a quirk to
a device with USB ID 0123:abcd treated as if it were 5678:beef.
This feature is useful to test an existing quirk, typically for a
newer model of the same vendor, without patching / rebuilding the
kernel driver.
The current implementation is fairly simplistic: since there is no API
for matching a usb_device_id to the given ID pair, it has an open code
to loop over the id table and matches only with vendor:product pair.
So far, this is OK, as all existing entries are with vendor:product
pairs, indeed. Once when we have another matching entry, however,
we'd need to update get_alias_quirk() as well.
Note that this option is provided only for testing / development. If
you want to have a proper support, contact to upstream for adding the
matching quirk in the driver code statically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the later change to allow a better
quirk ID management. In the current USB-audio code, there are a few
places looking at usb_device idVendor and idProduct fields directly
even though we have already a static member in snd_usb_audio.usb_id.
This patch modifies such codes to refer to the latter field.
For achieving this, two slightly intensive changes have been done:
- The snd_usb_audio object is set/reset via dev_getdrv() for the given
USB device; it's needed for minimizing the changes for some existing
quirks that take only usb_device object.
- __snd_usbmidi_create() is introduced to receive the pre-given usb_id
argument. The exported snd_usbmidi_create() is unchanged by calling
this new function internally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TEAC UD-501/UD-503/NT-503 fail to switch properly between different
rate/format. Similar to 'Playback Design', this patch corrects the
invalid clock source error for TEAC products and avoids complete
freeze of the usb interface of 503 series.
Signed-off-by: Guillaume Fougnies <guillaume@eulerian.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree in
anyway sooner or later.]
Core
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc
- A few code cleanups of fm801 driver
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Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[ NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree
in anyway sooner or later. ]
Core:
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC:
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio:
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire:
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio:
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc:
- A few code cleanups of fm801 driver"
* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
ALSA: timer: Code cleanup
ALSA: timer: Harden slave timer list handling
ALSA: hda - Add fixup for Dell Latitidue E6540
ALSA: timer: Fix race among timer ioctls
ALSA: hda - add codec support for Kabylake display audio codec
ALSA: timer: Fix double unlink of active_list
ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
ALSA: hda - fix the headset mic detection problem for a Dell laptop
ALSA: hda - Fix white noise on Dell Latitude E5550
ALSA: hda_intel: add card number to irq description
ALSA: seq: Fix race at timer setup and close
ALSA: seq: Fix missing NULL check at remove_events ioctl
ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
ASoC: AMD: Add missing include file
ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
ALSA: usb: Add native DSD support for Oppo HA-1
ASoC: Make aux_dev more like a generic component
ASoC: bcm2835: cleanup includes by ordering them alphabetically
ASoC: AMD: Manage ACP 2.x SRAM banks power
...
Pull trivial tree updates from Jiri Kosina.
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial:
floppy: make local variable non-static
exynos: fixes an incorrect header guard
dt-bindings: fixes some incorrect header guards
cpufreq-dt: correct dead link in documentation
cpufreq: ARM big LITTLE: correct dead link in documentation
treewide: Fix typos in printk
Documentation: filesystem: Fix typo in fs/eventfd.c
fs/super.c: use && instead of & for warn_on condition
Documentation: fix sysfs-ptp
lib: scatterlist: fix Kconfig description
The commit [da6d276957: ALSA: usb-audio: Add resume support for
Native Instruments controls] brought a regression where the Native
Instrument audio devices don't get the correct value at update due to
the missing shift at writing. This patch addresses it.
Fixes: da6d276957 ('ALSA: usb-audio: Add resume support for Native Instruments controls')
Reported-and-tested-by: Owen Williams <owilliams@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM may still have a leftover instance after disconnection and
it delays its release. The problem is that the PCM close code path of
USB-audio driver has a call of snd_usb_autosuspend(). This involves
with the call of usb_autopm_put_interface() and it may lead to a
kernel Oops due to the NULL object like:
BUG: unable to handle kernel NULL pointer dereference at 0000000000000190
IP: [<ffffffff815ae7ef>] usb_autopm_put_interface+0xf/0x30 PGD 0
Call Trace:
[<ffffffff8173bd94>] snd_usb_autosuspend+0x14/0x20
[<ffffffff817461bc>] snd_usb_pcm_close.isra.14+0x5c/0x90
[<ffffffff8174621f>] snd_usb_playback_close+0xf/0x20
[<ffffffff816ef58a>] snd_pcm_release_substream.part.36+0x3a/0x90
[<ffffffff816ef6b3>] snd_pcm_release+0xa3/0xb0
[<ffffffff816debb0>] snd_disconnect_release+0xd0/0xe0
[<ffffffff8114d417>] __fput+0x97/0x1d0
[<ffffffff8114d589>] ____fput+0x9/0x10
[<ffffffff8109e452>] task_work_run+0x72/0x90
[<ffffffff81088510>] do_exit+0x280/0xa80
[<ffffffff8108996a>] do_group_exit+0x3a/0xa0
[<ffffffff8109261f>] get_signal+0x1df/0x540
[<ffffffff81040903>] do_signal+0x23/0x620
[<ffffffff8114c128>] ? do_readv_writev+0x128/0x200
[<ffffffff810012e1>] prepare_exit_to_usermode+0x91/0xd0
[<ffffffff810013ba>] syscall_return_slowpath+0x9a/0x120
[<ffffffff817587cd>] ? __sys_recvmsg+0x5d/0x70
[<ffffffff810d2765>] ? ktime_get_ts64+0x45/0xe0
[<ffffffff8115dea0>] ? SyS_poll+0x60/0xf0
[<ffffffff818d2327>] int_ret_from_sys_call+0x25/0x8f
We have already a check of disconnection in snd_usb_autoresume(), but
the check is missing its counterpart. The fix is just to put the same
check in snd_usb_autosuspend(), too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset
but they use their own vendor ID.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.
Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.
However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.
Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).
Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.
v2: incorporated Takashi Iwai's suggestion for the quirk application
method
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb_protocol_ops structures are never modified, so declare them as
const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fix multiple spelling typos found in
various part of kernel.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The snd_rawmidi_global_ops structures are never modified, so declare them
as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 939f325f4a ("usb: add usb_endpoint_maxp() macro") and commit
29cc88979a ("USB: use usb_endpoint_maxp() instead of le16_to_cpu()")
introduced a new helper macro. This trivial patch convert remaining
users found in ua101 driver.
Signed-off-by: Cheah Kok Cheong <thrust73@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the many faults of the QinHeng CH345 USB MIDI interface chip is
that it does not handle received SysEx messages correctly -- every second
event packet has a wrong code index number, which is the one from the last
seen message, instead of 4. For example, the two messages "FE F0 01 02 03
04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event
packets:
correct: CH345:
0F FE 00 00 0F FE 00 00
04 F0 01 02 04 F0 01 02
04 03 04 05 0F 03 04 05
04 06 07 08 04 06 07 08
04 09 0A 0B 0F 09 0A 0B
04 0C 0D 0E 04 0C 0D 0E
05 F7 00 00 05 F7 00 00
A class-compliant driver must interpret an event packet with CIN 15 as
having a single data byte, so the other two bytes would be ignored. The
message received by the host would then be missing two bytes out of six;
in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7".
These corrupted SysEx event packages contain only data bytes, while the
CH345 uses event packets with a correct CIN value only for messages with
a status byte, so it is possible to distinguish between these two cases by
checking for the presence of this status byte.
(Other bugs in the CH345's input handling, such as the corruption resulting
from running status, cannot be worked around.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CH345 USB MIDI chip has two output ports. However, they are
multiplexed through one pin, and the number of ports cannot be reduced
even for hardware that implements only one connector, so for those
devices, data sent to either port ends up on the same hardware output.
This becomes a problem when both ports are used at the same time, as
longer MIDI commands (such as SysEx messages) are likely to be
interrupted by messages from the other port, and thus to get lost.
It would not be possible for the driver to detect how many ports the
device actually has, except that in practice, _all_ devices built with
the CH345 have only one port. So we can just ignore the device's
descriptors, and hardcode one output port.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.
This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.
It would benefit from some regresison testing with other devices if
possible.
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to verify that "value" is either zero or one, so we test if it
is greater than one. Unfortunately, this is a signed int so it could
also be negative. I think this is harmless but it introduces a static
checker warning. Let's make "value" unsigned.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of cval->cached should be zero-based (including master channel).
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In theory, the device may get suspended even at runtime PM suspend.
Currently we don't save the mixer state for autopm, and it may bring
inconsistency.
This patch removes the special handling for autosuspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent fix of runtime PM for USB-audio driver, we got a
lockdep warning like:
=============================================
[ INFO: possible recursive locking detected ]
4.2.0-rc8+ #61 Not tainted
---------------------------------------------
pulseaudio/980 is trying to acquire lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
but task is already holding lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
This comes from snd_usb_autoresume() invoking down_read() and it's
used in a nested way. Although it's basically safe, per se (as these
are read locks), it's better to reduce such spurious warnings.
The read lock is needed to guarantee the execution of "shutdown"
(cleanup at disconnection) task after all concurrent tasks are
finished. This can be implemented in another better way.
Also, the current check of chip->in_pm isn't good enough for
protecting the racy execution of multiple auto-resumes.
This patch rewrites the logic of snd_usb_autoresume() & co; namely,
- The recursive call of autopm is avoided by the new refcount,
chip->active. The chip->in_pm flag is removed accordingly.
- Instead of rwsem, another refcount, chip->usage_count, is introduced
for tracking the period to delay the shutdown procedure. At
the last clear of this refcount, wake_up() to the shutdown waiter is
called.
- The shutdown flag is replaced with shutdown atomic count; this is
for reducing the lock.
- Two new helpers are introduced to simplify the management of these
refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
the shutdown state, and does autoresume. snd_usb_unlock_shutdown()
does the opposite. Most of mixer and other codes just need this,
and simply returns an error if it receives an error from lock.
Fixes: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Gustard DAC-X20U.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input terminal parser recurses into the referenced clock entity to verify
it is existant and thus the terminal descriptor is valid. The actual property
values of the term instance which is initially parsed must not be overriden by
the recursion. For this to work the term properties have to be assigned after
recursing into the referenced clock entity descriptors.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix for deadlock in PM in commit [1ee23fe07e: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio Class version 2.0 supports three different parameter block sizes for
CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes
(5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter
Block). Use the correct size according to the specific control as it was
already done for UACv1. The allocated block size for control requests is
increased to support the 4 byte worst case.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2.
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a monitor stream is active, the next PCM stream access results in
EBUSY error because of the check in line6_stream_start(). Fix this by
just skipping the submission of pending URBs when the stream is
already running instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431
Cc: <stable@vger.kernel.org> # v4.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Steinberg MI2 and MI4 interfaces are compatible with the USB class
audio spec, but the MIDI part of the devices is reported as a vendor
specific interface.
This patch adds entries to quirks-table.h to recognize the MIDI
endpoints. Audio functionality was already working and is unaffected by
this change.
Signed-off-by: Dominic Sacré <dominic.sacre@gmx.de>
Signed-off-by: Albert Huitsing <albert@huitsing.nl>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver worked around an error in the MAYA44 USB(+)'s mixer unit
descriptor by aborting before parsing the missing field. However,
aborting parsing too early prevented parsing of the other units
connected to this unit, so the capture mixer controls would be missing.
Fix this by moving the check for this descriptor error after the parsing
of the unit's input pins.
Reported-by: nightmixes <nightmixes@gmail.com>
Tested-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer control names for the ESI Maya44 USB+ (which appears to be
identical width the AudioTrak Maya44 USB).
Reported-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk allows us to avoid the noisy:
current rate 0 is different from the runtime rate
message every time playback starts. While USB DAC in the RR2150
supports reading the sample rate, it never returns a sample rate
other than zero in my observation with common sample rates.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this from the logs:
usb 7-1: New USB device found, idVendor=046d, idProduct=08ca
...
usb 7-1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong.
usb 7-1: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by sparse:
sound/usb/bcd2000/bcd2000.c:73:1: warning: symbol 'devices_used' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam Studio (045e:0772) needs a similar quirk for
suppressing the wrong sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Roland SC-D70 reports its device class as vendor specific class and
the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output.
In the quirks table the sampling rate was hard-coded to 44100 Hz
and therefore not worked when the sound module was in 48000 Hz mode.
In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE
but as the sound module reports incorrect bSubframeSize in its
descriptors, additional change is made in format.c to detect it and
to override it (which uses the existing code for Edirol SD-90).
Tested both when the sound module was in 44100 Hz mode and 48000 Hz
mode and both audio input and output. MIDI related part of the driver
is not touched.
Signed-off-by: Takamichi Horikawa <takamichiho@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate.
Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed.
Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961.
Signed-off-by: Adam Honse <calcprogrammer1@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3237"
Signed-off-by: Dmitry M. Fedin <dmitry.fedin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this quirk allows us to avoid the noisy
"cannot get freq at ep 0x1" message in dmesg output every time
playback starts.
This ought to affect other Benchmark DAC1 variations using the same
"Microchip Technology, Inc." chip as well, but I have only tested
with the "Pre" variant.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device complies to the UAC1 standard but hides that fact with
proprietary descriptors. The autodetect quirk for Roland devices
catches the audio interface but misses the MIDI part, so a specific
quirk is needed.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-by: Rafa Lafuente <rafalafuente@gmail.com>
Tested-by: Raphaël Doursenaud <raphael@doursenaud.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usages of clamp() macro in sound/usb/line6/playback.c are just
wrong, the low and high values are swapped.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix problem where playback of Denon DA-300USB DAC sometimes does not
start and leads to error messages like "clock source 41 is not valid,
cannot use".
Solution: Treat this device the same as other Denon/Marantz devices in
sound/usb/quirks.c.
Tested with both PCM and DSD formats.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261
Signed-off-by: Frank C Guenther <bugzilla.frnkcg@spamgourmet.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate.
This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound.
[minor tidy up by tiwai]
Signed-off-by: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The address cannot be negative so make it unsigned. Also, an unsigned
int is always sufficient for the length, so no need to overdo it with a
size_t. Finally, add in range checks to see if the values passed in
actually fit where they are used.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmware version is a single byte so have the variable type agree.
Since the address to this member is passed to the read function, using
an int is not even portable.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put an upper bound on how long we will wait for the device to respond to
a read/write request (i.e., 100 milliseconds) and return an error if
this is reached.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device indicates the result of a read/write operation by making the
status available on a subsequent request from the driver. This is not
ready immediately, though, so the driver is currently slamming the
device with hundreds of pointless requests before getting the expected
response. Add a two millisecond delay before each attempt. This is
approximately the behavior observed with version 4.2.7.1 of the Windows
driver.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The first URBs are submitted during the prepare stage. When .trigger is
called, the ALSA core saves a trigger tstamp that doesn't correspond to
the actual time when the samples are submitted. The trigger_tstamp is
now updated when the first data are submitted to avoid any time offsets.
A usb-specific trigger_tstamp_pending_update flag is used for now,
at some point the flag would need to move to the ALSA core, USB
is not the only interface where silent block transfers are programmed
as part of the prepare stage, with actual data enabled when .trigger
is called.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a unique name for each driver instead of using "line6usb" for
all of them. This will allow for different configurations based on the
driver type.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is unlikely this function would ever be used in a context without a
pointer to a `struct usb_line6_toneport', so grab the device type from
it rather than having the caller do it.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a predicate for testing if the device supports source selection to
make the conditional logic around this a bit cleaner.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_start_timer passes an unsigned int as argument to be used in mod_timer
which is then used by mod_timer as unsigned long, this just fixes up the
argument type. This change helps make static code checkers happy.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is only an API consolidation and should make things more readable
it replaces var * HZ / 1000 by msecs_to_jiffies(var).
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use of this function ended with commits 3e58c868db ("staging: line6:
drop midi_mask_receive") and af89d2897a ("staging: line6: drop
midi_mask_transmit".)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function has not been used since merging the driver into the kernel
(and a good while before that.)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of them are rather relevant with the definitions in driver.h,
and there are only a few lines, so just rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just reformatting the comments and typos fixed, no functional
changes. Particularly,
- avoid the kerneldoc marker "/**",
- reduce multiple comment lines into single lines,
- corrected wrongly referred function names
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-space API definition for usb_stream stuff should be moved
to include/uapi/sound to be exposed publicly.
While we're at it, add the missing ifdef guard for double inclusion,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both playback and capture callbacks are identical, so let's merge
them.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code deals with the stream start / stop solely via
line6_pcm_acquire() and line6_pcm_release(). This was (supposedly)
intended to avoid the races, but it doesn't work as expected. The
concurrent acquire and release calls can be performed without proper
protections, thus this might result in memory corruption.
Furthermore, we can't take a mutex to protect the whole function
because it can be called from the PCM trigger callback that is an
atomic context. Also spinlock isn't appropriate because the function
allocates with kmalloc with GFP_KERNEL. That is, these function just
lead to singular problems.
This is an attempt to reduce the existing races. First off, separate
both the stream buffer management and the stream URB management. The
former is protected via a newly introduced state_mutex while the
latter is protected via each line6_pcm_stream lock.
Secondly, the stream state are now managed in opened and running bit
flags of each line6_pcm_stream. Not only this a bit clearer than
previous combined bit flags, this also gives a better abstraction.
These rewrites allows us to make common hw_params and hw_free
callbacks for both playback and capture directions.
For the monitor and impulse operations, still line6_pcm_acquire() and
line6_pcm_release() are used. They call internally the corresponding
functions for both playback and capture streams with proper lock or
mutex. Unlike the previous versions, these function don't take the
bit masks but the only single type value. Also they are supposed to
be applied only as duplex operations.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing prev_fsize in line6_pcm_acquire() is pretty racy.
This can be called at any time while the stream is being played.
Rather better to clear prev_fbuf and prev_fsize at the proper place
like the stream stop for capture, and just after copying the monitor /
impulse data inside the spinlock.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The impulse and monitor handling in submit_audio_out_urb() isn't
protected thus this can be racy with the capture stream handling.
This patch extends the range to protect via each stream's spinlock
(now the whole submit_audio_*_urb() are covered), and take the capture
stream lock additionally for the impulse and monitor handling part.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check of multi configurations before snd_card_new() as a
short path, and reduce superfluous pointer references.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the private data individually in each driver's
probe at first, let snd_card_new() allocate the data that is called in
line6_probe(). This simplifies the primary probe functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface argument is used just for retrieving the assigned
device, which can be already found in line6->ifcdev. Drop them from
the callbacks. Also, pass the usb id to private_init so that the
driver can deal with it there. This is a preliminary work for the
further cleanup to move the whole allocation into driver.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A minor optimization; while pausing, the driver just copies the zero
that doesn't need any volume changes.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM stream buffer allocation and free are identical for both
playback and capture streams. Provide single helper functions.
These are used only in pcm.c, thus they can be even static.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to unlink and sync URBs are identical for both playback and
capture streams. Consolidate to single helper functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new line6_pcm_stream structure and group individual
fields of snd_line6_pcm struct to playback and capture groups.
This patch itself just does rename and nothing else. More
meaningful cleanups based on these fields shuffling will follow.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the problem still really remains, we should fix it instead of
papering over it like this...
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a decremental loop without particular reasons worsens the
readability a lot. Use incremental loops instead.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The trigger callback is already spinlocked, so we need no more lock
here (even for the linked substreams). Let's drop it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_pcm_acquire() tries to restore the newly obtained resources at
the error path. But some flags aren't recorded and released properly
when the corresponding buffer is already present. These bits have to
be cleared in the error recovery, too.
Also, "flags_final" can be initialized to zero since we pass only the
subset of "channels" bits.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a few places using bits OR wrongly for condition checks.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The midi_transmit_lock is used always inside the send_urb_lock, thus
it doesn't play any role. Let's kill it. Also, rename
"send_urb_lock" as a more simple name "lock" since this is the only
lock for midi.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function isn't used any longer after rewriting from sysfs to leds
class in toneport.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix memory leak at probe error path by rearranging the call order in
line6_destruct() so that the common destructor is always called.
Also this simplifies the error path to a single goto label.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of non-standard sysfs, reimplement the LED controls on
TonePort as LED class devices.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently disconnect callback is used as a driver's destructor, and
this has to be called not only at the disconnection time but also at
the error paths during probe.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that timer_del_sync() in the destructor can be called safely at
any time. Also move the mod_timer() call in toneport_setup(), which
is a bit clearer place.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface and driver objects are always set when callbacks are
called. Drop such superfluous NULL checks in init and disconnect
calls of each driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's utterly unsafe to proceed further the disconnect procedure if the
assigned usbdev is inconsistent with the expected object. Better to
put a WARN_ON() for more cautions and abort immediately.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver version string was removed in an ealier commit for being
useless. These are equally useless.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct spelling includes the space. Fix this in strings and
comments that refer to the manufacturer.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call line6_pcm_disconnect() at disconnect to make sure that all URBs
are cleared. Also reduce the superfluous snd_pcm_stop() calls from
the function (and remove the unused function) since the streams are
guaranteed to be stopped at this point via snd_card_disconnect().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Calling line6_pcm_disconnect() at suspend callback is superfluous and
rather confusing. Let's get rid of it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Such a debug is needed in the core code, not in each lowlevel driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is rather useless for a driver that has been already merged into
the official tree.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fairly big rewrite regarding the card resource management in
line6 drivers:
- The card creation is moved into line6_probe(). This adds the global
destructor to private_free, so that each driver doesn't have to call
it any longer.
- The USB disconnect callback handles the card release, thus each
driver needs to concentrate on only its own resources. No need to
snd_card_*() call in the destructor.
- Fix the potential stall in disconnection by removing
snd_card_free(). It's replaced with snd_card_free_when_closed()
for asynchronous release.
- The only remaining operation for the card in each driver is the call
of snd_card_register(). All the rest are dealt in the common module
by itself.
- These ended up with removal of audio.[ch] as a result of a reduction
of one layer. Each driver just needs to call line6_probe().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM trigger callback is guaranteed to be called already in
spinlock / irq-disabled context.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The line6 drivers don't support the full resume although they set
SNDRV_PCM_INFO_RESUME. These flags have to be dropped to inform
properly to the user-space.
Also, drop the CONFIG_PM in trigger callbacks, too, which are rather
superfluous.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous fix for PCM, attach the card-specific resource into
rawmidi->private_data instead of handling in a snd_device object.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of handling the card-specific resource in snd_device, attach
it into pcm->private_data and release it directly in private_free.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of sysfs and the conditional build with Kconfig, implement the
handling of the impulse response controls via control API, and always
enable the build. Two new controls, "Impulse Response Volume" and
"Impulse Response Period" are added as a replacement for the former
sysfs files.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split to each individual driver for POD, PODHD, TonePort and Variax
with a core LINE6 helper module. The new modules follow the standard
ALSA naming rule with snd prefix: snd-usb-pod, snd-usb-podhd,
snd-usb-toneport and snd-usb-variax, together with the corresponding
CONFIG_SND_USB_* Kconfig items.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
