"Equalizer Function" ctl in wm8983 codec driver is enum, while the
current driver accesses wrongly via value.integer.value[]. They have
to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"MBC Mode", "VSS Mode", "VSS HPF Mode" and "Enhanced EQ Mode" ctls in
wm8958 codec driver are enum, while the current driver accesses
wrongly via value.integer.value[]. They have to be via
value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
"DRC Mode" and "EQ Mode" ctls in wm8904 codec driver are enum, while
the current driver accesses wrongly via value.integer.value[]. They
have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"DAI Mode" ctl in wm8753 codec driver is enum, while the current
driver accesses wrongly via value.integer.value[]. They have to be
via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Codec Mode" and "Audio Switch" ctls in wl1273 codec driver are enum,
while the current driver accesses wrongly via value.integer.value[].
They have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"FIFO Mode" ctl in tlv320dac33 codec driver is enum, while the current
driver accesses wrongly via value.integer.value[]. They have to be
via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Biquad1 Mode" and "Biquad2 Mode" ctls in max98095 codec driver are
enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"EQ1 Mode" and "EQ2 Mode" ctls in max98088 codec driver are enum,
while the current driver accesses wrongly via value.integer.value[].
They have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Sidetone Status" and "ANC Status" ctls in ab8500 codec driver are
enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"DAC1 High Pass Filter Mode" & co in da732x codec driver are enum,
while the current driver accesses wrongly via value.integer.value[].
They have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"PCM channel mixer" ctl in cs42l51 codec driver is enum, while the
current driver accesses wrongly via value.integer.value[]. They have
to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Playback Switch" and "Lineout Mux" ctls in medfld machine driver are
enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Speaker Function", "Input Select" and "Jack Function" ctls in rx51
driver are enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Jack Function", "Speaker Function" and "Input Select" ctls in n810
driver are enum, while the current driver accesses wrongly via
value.integer.value[]. They have to be via value.enumerated.item[]
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Jack Function" and "Speaker Function" ctls in tosa are enum, while
the current driver accesses wrongly via value.integer.value[]. They
have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Jack Function" and "Speaker Function" ctls in spitz are enum, while
the current driver accesses wrongly via value.integer.value[]. They
have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Jack Function" and "Speaker Function" ctls in poodle are enum, while
the current driver accesses wrongly via value.integer.value[]. They
have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Input Select" ctl in magician driver is an enum, while the current
driver accesses wrongly via value.integer.value[]. They have to be
via value.enumerated.item[] instead.
(Meanwhile "Headphone Switch" and "Speaker Switch" are boolean, so
they should stay to access via value.integer.value[] as is.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
"Jack Function" and "Speaker Function" ctls in corgi are enum, while
the current driver accesses wrongly via value.integer.value[]. They
have to be via value.enumerated.item[] instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dapm_dai_link_get() and _put() access the associated ctl
values as value.integer.value[]. However, this is an enum ctl, and it
has to be accessed via value.enumerated.item[]. The former is long
while the latter is unsigned int, so they don't align.
Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters')
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The variable cmd_id is only assigned some value and is never used.
Signed-off-by: Sudip Mukherjee <sudip.mukherjee@codethink.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
HDSPM driver contains a code issuing zero-division potentially in
system sample rate ctl code. This patch fixes it by not processing
a zero or invalid rate value as a divisor, as well as excluding the
invalid value to be passed via the given ctl element.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While registering pardev, the irq_func was also registered. As a
result when we tried to probe for the card, an interrupt was generated
and in the ISR we tried to dereference private_data. But private_data
is still NULL as we have not yet done snd_mts64_create(). Lets probe
for the card after mts64 is created.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Fixes: 94a573500d48 ("ALSA: mts64: use new parport device model")
Signed-off-by: Sudip Mukherjee <sudip.mukherjee@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
X32 ABI takes the 64bit timespec, thus the timer user status ioctl becomes
incompatible with IA32. This results in NOTTY error when the ioctl is
issued.
Meanwhile, this struct in X32 is essentially identical with the one in
X86-64, so we can just bypassing to the existing code for this
specific compat ioctl.
Cc: <stable@vger.kernel.org> # v3.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer user status compat ioctl returned the bogus struct used for
64bit architectures instead of the 32bit one. This patch addresses
it to return the proper struct.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous fixes for ctl and PCM, we need a fix for
incompatible X32 ABI regarding the rawmidi: namely, struct
snd_rawmidi_status has the timespec, and the size and the alignment on
X32 differ from IA32.
This patch fixes the incompatible ioctl for X32.
Cc: <stable@vger.kernel.org> # v3.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
X32 ABI uses the 64bit timespec in addition to 64bit alignment of
64bit values. This leads to incompatibilities in some PCM ioctls
involved with snd_pcm_channel_info, snd_pcm_status and
snd_pcm_sync_ptr structs. Fix the PCM compat ABI for these ioctls
like the previous commit for ctl API.
Reported-by: Steven Newbury <steve@snewbury.org.uk>
Cc: <stable@vger.kernel.org> # v3.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The X32 ABI takes the same alignment like x86-64, and this may result
in the incompatible struct size from ia32. Unfortunately, we hit this
in some control ABI: struct snd_ctl_elem_value differs between them
due to the position of 64bit variable array. This ends up with the
unknown ioctl (ENOTTY) error.
The fix is to add the compat entries for the new aligned struct.
Reported-and-tested-by: Steven Newbury <steve@snewbury.org.uk>
Cc: <stable@vger.kernel.org> # v3.4+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit drops implementation of duplex streams synchronization
from ALSA dice driver, due to a reason of hardware design. This patch
allows dice-based units to generate sounds correctly when isochronous
packet streaming starts at first time.
In IEC 61883-6:2005, CIP packetization layer for AM824 data format
utilizes the value of SYT field in CIP header of received packet for
a reference to phase lock loop. Figure 3 in clause 4.3 describes it.
The value is an offset from cycle_time field of every cycle start packet
from cycle master on IEEE 1394 bus. The time calculated with these two
fields is called as 'presentation timestamp' which represents the time
to play data included in the packet.
Although, this idea includes some problems due to accuracy of timekeep in
cycle master, accuracy of transmission of cycle start packet on the bus
with the other units, accuracy of sampling clock in data transmitter side
and accuracy of replay in data receiver side. In most case, these
accuracies somewhat worse because there's no such ideal hardwares in this
world.
For the issues, ASICs for Dice include Jitter Elimination Technologies
(JET) PLL. The PLL can handle several sources of clock and compensate it
with high-precision internal clock source. The sequence of value in syt
field of received AMDTP packets is one of the sources, therefore
transmitters on IEEE 1394 bus should transfer it.
On the other hand, current ALSA dice driver is programmed with a mode of
duplex streams with synchronization. In this mode, the driver outputs
packets after some incoming packets are handled, to re-use the value of
SYT field in incoming packets to the value for outgoing packets. This mode
is enabled when source signal of sampling clock is set to internal, and
this is a major use case. Thus, in most cases, the unit receives no packets
during a short time after packet streaming starts.
As long as I experienced, this causes the units to generate no sounds at
first time to receive packets. This issue occurs only with Dice II. I guess
this is due to a quirk of the PLL. In short, the PLL cannot generate firm
signals to ADCs/DACs or the other ICs when no packets are received in the
beginning of packet streaming. While, on second time or later, the unit
generates sound correctly. I guess that starting packet streaming at first
time sets the PLL correctly.
Well, still based on my hypothesis and no way to prove it, this commit
drops duplex streams synchronization from this driver. At least, the PLL
requires the sequence of value in SYT field of received AMDTP packets as
one of source of clock signals with internal clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the interrupt handler of HD-audio driver assumes that no irq
update is needed while processing the irq. But in reality, it has
been confirmed that the HW irq is issued even during the irq
handling. Since we clear the irq status at the beginning, process the
interrupt, then exits from the handler, the lately issued interrupt is
left untouched without being properly processed.
This patch changes the interrupt handler code to loop over the
check-and-process. The handler tries repeatedly as long as the IRQ
status are turned on, and either stream or CORB/RIRB is handled.
For checking the stream handling, snd_hdac_bus_handle_stream_irq()
returns a value indicating the stream indices bits. Other than that,
the change is only in the irq handler itself.
Reported-by: Libin Yang <libin.yang@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4271 has three power domains: vd, vl and va.
Enable them all, as long as the codec is in use.
While at it, factored out the reset code into its own function.
Signed-off-by: Pascal Huerst <pascal.huerst@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add these widgets to allow another path from I2S input.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We cannot use strcpy() to write to a const char * location. This is
causing a 'BUG: unable to handle kernel paging request' error at boot
when using the cht-bsw-rt5645 driver.
With this patch we also fix a wrong indexing in the driver where the
codec_name of the wrong dai_link is being overwritten.
Signed-off-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To improve I2S flow, this patch removes soft reset and adds second
I2S clock to use.
Signed-off-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As pointed out by Zhangfei Gao, the sspa_div variable in
brownstone_wm8994_hw_params() is completely unused, so as a cleanup
following a prior patch, this removes both the variable and the division.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Headphone needs enough delay time before unmuting for avoiding pop sound.
We extend the delay time to make sure headphone doesn't pop.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When playing audio through headphone, headphone makes pop noise if system
resumes from S3 to S0. We modify the sequence of writing register for
avoiding pop sound.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Smatch complains that we might reach the end of this loop without
finding what we're looking for leading to a buffer overflow.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HP EliteBook 755 G2 with ALC3228 (ALC280) codec [103c:221c] requires
the known fixup (ALC269_FIXUP_HEADSET_MIC) for making the headset mic
working. Also, it suffers from the loopback noise problem, so we
should disable aamix path as well.
Reported-by: Derick Eddington <derick.eddington@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On one of the machines we enable, we found that the actual speaker volume
did not always correspond to the volume set in alsamixer. This patch
fixes that problem.
This patch was orginally written by Kailang @ Realtek, I've rebased it
to fit sound git master.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1549660
Co-Authored-By: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After login to the desktop on Dell Inspiron 3162,
there's a very loud background noise comes from the builtin speaker.
The noise does not go away even if the speaker is muted.
The noise disappears after using the aamix fixup.
Codec: Realtek ALC3234
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0255
Subsystem Id: 0x10280725
Revision Id: 0x100002
No Modem Function Group found
BugLink: http://bugs.launchpad.net/bugs/1549620
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a previous commit, ALSA oxfw driver retries transferring MIDI
messages at transaction failure for scs1x. On the other hand, there're
fatal transaction error. Then, no MIDI messages reach to the unit anymore.
In this case, MIDI substream should be terminated.
This commit stops MIDI transmission after the fatal error occurs.
Unfortunately, unlike ALSA PCM functionality, ALSA rawmidi core has no
feature to discontinue MIDI substream runtime in kernel side, thus this
commit just stops MIDI transmission without notifying it to userspace.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, ALSA oxfw driver has a TODO to retry MIDI transferring
at transaction failure.
This commit achieves it. Current implementation uses snd_rawmidi_transmit()
to transfer messages, thus the target MIDI messages are not in buffer when
transaction failure is detected. Although we cannot use a pair of
snd_rawmidi_transmit_peek() and snd_ramwidi_transmit_ack(), the
messages are still in scs1x specific structure and the data is available
for retries.
This commit adds a member to the structure for the length of buffered
messages, and uses the value again at retries.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
