This commit takes ALSA oxfw driver to perform sequence replay for media
clock recovery. Unfortunately, OXFW970 ASIC and its firmware has a quirk
called jumbo payload which skips several isochronous cycles for packet
transmission, thus the sequence replay is just adopted to OXFW971 ASIC.
As well as Fireworks, OXFW ASICs also ignores presentation time against
the way in IEC 61883-1/6.
The sequence replay is tested with below models:
* Tascam FireOne
* Stanton Magnetics SCS.1m
* Apogee Duet FireWire
For below models, the sequence replay is tested to be disabled:
* Griffin FireWave
* Behringer F-Control Audio 202
* Loud Technology Tapco Link.FireWire 4x6
* Loud Technology Mackie Onyx Satellite
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Echo Digital Audio Corporation had US patent US7599388B2 titled as
'System and method for high-bandwidth serial bus data transfer'. In the
patent, dual-banked shared memory is used to deliver data between
serial bus transmission and processor in FIFO way. The patent seems to be
used for Fireworks board module. The mechanism is not compliant to
synchronization based on presentation time expressed in syt field
of CIP header. Fireworks board module takes care of the sequence of
the number of data blocks per packet and just ignores the value of syt
field.
This commit takes fireworks driver to performs sequence replay for media
clock recovery. As long as I tested, Audiofire 2 and 4 have a quirk to
skip an isochronous cycle several thousands after starting packet
transmission.
The sequence replay is tested with below models:
* Loud Technology Mackie 400f
* Echo Audio Audiofire 12 (DSP model)
* Echo Audio Audiofire 12 (FPGA model)
* Echo Audio Audiofire 8 (DSP model)
* Echo Audio Audiofire 8 (FPGA model)
* Echo Audio Audiofire Pre8
* Echo Audio Audiofire 4
* Echo Audio Audiofire 2
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the design of Fireworks board module, the device does't adjust its
media clock voluntarily by the sequence of presentation time expressed in
syt field of CIP header of received packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit to drop the HDA-specific mute-LED control,
e65bf99718 ("ALSA: HDA - remove the custom implementation for the
audio LED trigger"), caused a regression on the mixer element read for
"Capture Switch" when it's built from bind controls. The function
create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly
via snd_hda_gen_add_kctl() instead of add_control(). Although the
commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in
add_control() as default, this code path fell out from the radar. As
a result, now the driver gives -EPERM error because of the lack of the
proper access bit at reading "Capture Switch" element value.
Fix the regression by setting the access bit properly.
Fixes: e65bf99718 ("ALSA: HDA - remove the custom implementation for the audio LED trigger")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634
Link: https://lore.kernel.org/r/20210531180633.27831-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add HD Audio PCI ID for Intel AlderLake-M. Add rules to
snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with
PCH-DMIC or Soundwire codecs, and legacy driver for the rest.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR,
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20210526121828.8460-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers of ALSA firewire stack can process packets for IT/IR context in
process context when the process operates ALSA PCM character device by
calling ioctl(2) with some requests. The ioctl requests are:
* SNDRV_PCM_IOCTL_HWSYNC
* SNDRV_PCM_IOCTL_SYNC_PTR
* SNDRV_PCM_IOCTL_REWIND
* SNDRV_PCM_IOCTL_FORWARD
* SNDRV_PCM_IOCTL_WRITEI_FRAMES
* SNDRV_PCM_IOCTL_READI_FRAMES
* SNDRV_PCM_IOCTL_WRITEN_FRAMES
* SNDRV_PCM_IOCTL_READN_FRAMES
This means that general application can process PCM frames apart from
hardware IRQ invocation, even if they are programmed by either IRQ-based
scheduling model or Timer-based scheduling model.
This commit add support for Timer-based scheduling model by allowing
PCM runtime to suppress both process wakeup per period and scheduling
hardware IRQ.
SNDRV_PCM_INFO_BATCH is obsoleted since ALSA IEC 61883-1/6 packet streaming
engine can report the number of transferred PCM frames within PCM period
boundary. The granularity equals to SYT_INTERVAL in blocking transmission.
In non-blocking transmission, it doesn't equal to SYT_INTERVAL but doesn't
exceed.
This patch is tested with PulseAudio, and --sched-model option of axfer
with fix against the issue reported at:
* https://lore.kernel.org/alsa-devel/687f9871-7484-1370-04d1-9c968e86f72b@linux.intel.com/#r
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527123253.174315-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Models in below series start transmission of packet after receiving the
sequence of packets:
* Digidesign Digi00x family
* RME Fireface series
Additionally, models in Tascam FireWire series start multiplexing PCM
frames into packets enough after receiving packets. It's required to
transfer packets on-the-fly for the above models according to nominal
sampling transfer frequency before starting sequence replay.
This commit allows drivers to decide whether the engine transfers packet
on-the-fly or not.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA IEC 61883-1/6 packet streaming engine uses pre-computed parameters
ideal for nominal sampling transfer frequency (STF) to transfer packets
to device since it was added 2011. As a result of user experience for a
decade, it is clear that the sequence is not suitable to some actual
devices. It takes the devices to generate noise, and causes any type of
discontinuity in the series of packet transferred from the device. It's
required for the engine to transfer packets according to effective STF.
The effective STF is given by media clock recovered by the sequence of
packet transferred from the target device. In the previous commit, the
sequence is already cached. The media clock recovery can be achieved by
analyzing the sequence.
In technological world, many ideas are proposed for media clock recovery.
However, the small part of them could be actually adopted in our case
since floating point arithmetic is not mostly available in Linux kernel
land.
This commit adopts the simple way from them; sequence replay, which means
that the sequence of parameters from incoming packet is used as is to
transfer outgoing packets. The media clock is not computed internally,
but the sequence of outgoing packet superficially looks to be generated by
the media clock.
The association between source and destination is decided when starting
AMDTP domain. When the target device supports a pair of isochronous packet
streams, the tx stream is source and the rx stream is destination. When it
supports two pair of streams, each of tx stream is associated to
corresponding rx stream in its order. When it supports less number of tx
streams than rx streams, the fist tx stream is selected for all of rx
streams. When it supports more tx streams than rx streams, the first tx
packet is associated to the rx stream.
As I noted in previous commit, the sequence of parameters from incoming
packet is different between devices, time to time. It is worse idea to
replay the sequence of parameters from a device for the sequence of
packet to the other devices even if they are in the same category of
device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In design of audio and music unit in IEEE 1394 bus, feedback of
effective sampling transfer frequency (STF) is delivered by packets
transferred from device. The devices supported by ALSA firewire stack
are categorized to three groups regarding to it.
* Group 1:
* Echo Audio Fireworks board module
* Oxford Semiconductor OXFW971 ASIC
* Digidesign Digi00x family
* Tascam FireWire series
* RME Fireface series
* Group 2:
* BridgeCo. DM1000/DM1100/DM1500 ASICs for BeBoB solution
* TC Applied Technologies DICE ASICs
* Group 3:
* Mark of the Unicord FireWire series
In group 1, the effective STF is determined by the sequence of the number
of events per packet. In group 2, the sequence of presentation timestamp
expressed in syt field of CIP header is interpreted as well. In group 3,
the presentation timestamp is expressed in source packet header (SPH) of
each data block.
I note that some models doesn't take care of effective STF with large
internal buffer. It's reasonable to name it as group 0:
* Group 0
* Oxford Semiconductor OXFW970 ASIC
The effective STF is known to be slightly different from nominal STF for
all of devices, and to be different between the devices. Furthermore, the
effective STF is known to be shifted for long-period transmission. This
makes it hard for software to satisfy the effective STF when processing
packets to the device.
The effective STF is deterministic as a result of analyzing the batch of
packet transferred from the device. For the analysis, caching the sequence
of parameter in the packet is required.
This commit adds an option so that AMDTP domain structure takes AMDTP
stream structure to cache the sequence of parameters in packet transferred
from the device. The parameters are offset ticks of syt field against the
cycle to receive the packet and the number of data blocks per packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't populate the const array dsp_dma_stream_ids the stack but instead
make it static. Makes the object code smaller by 21 bytes.
Before:
text data bss dec hex filename
189012 70376 192 259580 3f5fc ./sound/pci/hda/patch_ca0132.o
After:
text data bss dec hex filename
188927 70440 192 259559 3f5e7 ./sound/pci/hda/patch_ca0132.o
(gcc version 10.3.0)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210526160616.3764119-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse throws the following warning:
sound/pci/lx6464es/lx_core.c:677:34: error: self-comparison always
evaluates to false
This comparison and error message make no sense, let's remove them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse throws the following warning:
sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always
evaluates to false
This is likely a 16+ year old confusion between vp2 and vp.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP ZBook Studio 17.3 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-4-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP ZBook Fury 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-3-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP ZBook Studio 15.6 Inch G8 is using ALC285 codec which is
using 0x04 to control mute LED and 0x01 to control micmute LED.
In the other hand, there is no output from right channel of speaker.
Therefore, add a quirk to make it works.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-2-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs
ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the
fixup, the mute/micmute LEDs work good.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210519170357.58410-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces
bug plock/pop noise over line out (green jack on the back) which can be
fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine.
Convert the existing entry for the same SSID to chain to apply this fixup
as well.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210524203726.2278-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR,
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210523071109.28940-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20210524120007.39728-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pm_runtime_get_sync will increment pm usage counter even it failed.
Forgetting to putting operation will result in reference leak here.
Fix it by replacing it with pm_runtime_resume_and_get to keep usage
counter balanced.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Yufen Yu <yuyufen@huawei.com>
Link: https://lore.kernel.org/r/20210524093811.612302-1-yuyufen@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of fixes that have come in since the merge window, mainly
device specific things. The fixes to the generic cards from
Morimoto-san are handling regressions that were introduced in the merge
window on at least the Kontron sl28-var3-ads2.
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Merge tag 'asoc-fix-v5.13-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.13
A collection of fixes that have come in since the merge window, mainly
device specific things. The fixes to the generic cards from
Morimoto-san are handling regressions that were introduced in the merge
window on at least the Kontron sl28-var3-ads2.
In former commit, ALSA IEC 61883-1/6 packet streaming engine drops
initial tx packets till the packet includes any event. This allows ALSA
bebob driver not to give option to skip initial packet since the engine
does drop the initial packet.
However, M-Audio ProFire Lightbridge has a quirk to stop packet
transmission after start multiplexing event to the packet. After several
thousands cycles, it restart packet transmission again.
This commit specializes the usage of initial skip option for the model.
Additionally, this commit expands timeout enough to wait processing
content of tx packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The member of callbacked in AMDTP stream structure is not used anymore.
Instead, ready_processing member is used to wake up yielding task of user
process.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices based on BeBoB ASICs or the devices in Tascam FireWire
series transfer a batch of NODATA packet or empty packet in the beginning
of packet streaming. To avoid processing them, current implementation uses
an option to skip processing content of tx packet during some initial
cycles. However, the hard-coded number is not enough useful.
This commit drops content of packets till the packet includes any event
firstly. The function of option is to skip processing content of tx packet
with any event after dropping.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The normal PCM operations are already blocked during the card power
off state in the PCM common ioctl handler, but the release isn't
covered. As the PCM stream release may also access the hardware,
let's block the release until the card power turns on.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210523090920.15345-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power_state argument of snd_power_wait() is superfluous, receiving
only SNDRV_POWER_STATE_D0. Let's drop it in all callers for
simplicity.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210523090920.15345-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Long long time ago, before the proper PM framework was introduced, it
was still possible to reach SNDRV_CTL_IOCTL_POWER ioctl during the
power off state. This ioctl existed as a main control for the suspend
resume state in the past, but the feature was already dropped along
with the standard PM framework. Now the read part,
SNDRV_IOCTL_POWER_STATE ioctl, returns practically always D0, and we
can do some minor optimization there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210523090920.15345-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we have more fine-grained power controls in each kcontrol ops, the
coarse checks of snd_power_wait() in a few control ioctls became
superfluous. Let's drop them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210523090920.15345-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the power state check is performed in various places (e.g. at
the entrance of quite a few ioctls), there can be still some pending
tasks that already went into the ioctl handler or other ops, and those
may access the hardware even after the power state check. For
example, kcontrol access ioctl paths that call info/get/put callbacks
may update the hardware registers. If a system wants to assure the
free from such hw access (like the case of PCI rescan feature we're
going to implement in future), this situation must be avoided, and we
have to sync such in-flight tasks finishing beforehand.
For that purpose, this patch introduces a few new things in core code:
- A refcount, power_ref, and a wait queue, power_ref_sleep, to the
card object
- A few new helpers, snd_power_ref(), snd_power_unref(),
snd_power_ref_and_wait(), and snd_power_sync_ref()
In the code paths that call kctl info/read/write/tlv ops, we check the
power state with the newly introduced snd_power_ref_and_wait(). This
function also takes the card.power_ref refcount for tracking this
in-flight task. Once after the access finishes, snd_power_unref() is
called to released the refcount in return. So the driver can sync via
snd_power_sync_ref() assuring that all in-flight tasks have been
finished.
As of this patch, snd_power_sync_ref() is called only at
snd_card_disconnect(), but it'll be used in other places in future.
Note that atomic_t is used for power_ref intentionally instead of
refcount_t. It's because of the design of refcount_t type; refcount_t
cannot be zero-based, and it cannot do dec_and_test() call for
multiple times, hence it's not suitable for our purpose.
Also, this patch changes snd_power_wait() to accept only
SNDRV_CTL_POWER_D0, which is the only value that makes sense.
In later patch, the snd_power_wait() calls will be cleaned up.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210523090920.15345-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need proper barriers to handle the power state change of the card
from different CPUs.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210523090920.15345-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/mixer_scarlett_gen2.c:2000:5: warning: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static?
Fixes: 265d1a90e4 ("ALSA: usb-audio: scarlett2: Improve driver startup messages")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20210522180900.GA83915@f59a3af2f1d9
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Yoga 9i has bass speaker, but the bass speaker can't work,
that is because there is an i2s amplifier on that speaker, need to
run ideapad_s740_coef() to initialize the amplifier.
And also needs to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK to rename
the speaker's mixer control name, otherwise the PA can't handle them.
BugLink: http://bugs.launchpad.net/bugs/1926165
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210522042645.14221-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some ASUS and MSI machines, the audio codec is alc1220 and the
Headphone is connected to audio mixer 0xf and DAC 0x5, in theory
the Headphone volume is controlled by DAC 0x5 (Heapdhone Playback
Volume), but somehow it is controlled by DAC 0x2 (Front Playback
Volume), maybe this is a defect on the codec alc1220.
Because of this issue, the PA couldn't switch the headphone and
Lineout correctly, If we apply the quirk CLEVO_P950 to those machines,
the Lineout and Headphone will share the audio mixer 0xc and DAC 0x2,
and generate Headphone+LO mixer, then PA could handle them when
switching between them.
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1206
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210522034741.13415-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, transfer delay is used in both IR and IT contexts. This
commit refactors regardless of transfer delay.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210522013303.49596-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current implementation pools the sequence in AMDTP domain. This is convenient
regarding to memory usage and computation time, however inconvenient for the
devices such that several rx streams are expected to transfer timing
information independently.
This commit refactors to pool the sequence per rx packet stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210522013303.49596-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many devices are unaware of syt field in rx CIP for playback timing.
This commit adds a flag to cancel processing syt field. Actually,
syt calculation is required to decide the number of events per rx packet.
The flag put 0xffff to CIP header of rx packet. On the other hand,
The value of syt field in CIP header of tx packet is unavailable. The
sequence of packet descriptor for tx packet includes 0 for the offset
of syt field to avoid computation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210522013303.49596-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of error is just done for the case that CIP header is available.
This commit moves auto variable into the branch to process CIP header.
Addresses-Coverity: ("Uninitialized scalar variable")
Fixes: c09010eeb3 ("ALSA: firewire-lib: handle the case that empty isochronous packet payload for CIP")
Suggested-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210520130409.GA170303@workstation
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.
Use info level instead of error for notifications.
Display the VID/PID so device_setup is targeted to the right device.
Display "enabled" message to easily confirm that the driver is loaded.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use usb_rcvctrlpipe() not usb_sndctrlpipe() for USB control input in
the Scarlett Gen 2 mixer driver. This fixes the device hang during
initialisation when used with the ehci-pci host driver.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/66a3d05dac325d5b53e4930578e143cef1f50dbe.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The direction of the pipe argument must match the request-type direction
bit or control requests may fail depending on the host-controller-driver
implementation.
Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe().
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: stable@vger.kernel.org # 5.10
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20210521133742.18098-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cur variable indicating the currently selected clock source can be
theoretically used as uninitialized after the recent commit
481f17c418 ("ALSA: usb-audio: Handle error for the current selector
gracefully"). For addressing it, initialize it before use.
Also, one place seems setting 0 to a wrong variable ret, instead of
cur; otherwise it makes little sense. Since the initialization is
done beforehand, we can get rid of this line, too.
Fixes: 481f17c418 ("ALSA: usb-audio: Handle error for the current selector gracefully")
Reported-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/4b261d68-f53f-240d-2d8a-2f88b337849d@canonical.com
Link: https://lore.kernel.org/r/s5hfsyhh97t.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
