A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:
- Set up the proper driver caps for this controller, similar as the
other AMD controller.
- Correct the DMA position reporting with the fixed FIFO size, which
is similar like as workaround used for VIA chip set.
- Even after the position correction, PulseAudio still shows
mysterious stalls of playback streams when a capture is triggered in
timer-scheduled mode. Since we have no clear way to eliminate the
stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
a temporary workaround.
This patch implements the workarounds. For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.
Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.
To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.
Fixes: a91c3fb2f8 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit bfcba288b9 ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object. This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.
Fix the bug by setting the PCM hw info flag locally.
Fixes: bfcba288b9 ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Pioneer DDJ-SX3 is a plain 12 32bit channel out and 10 channel in
PCM/midi controller. The PCM part is "vendor specific".
It needs the "ignore invalid bsynchaddress" patch as it uses 0 for that.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cards have alternate setting with non-PCM format as the first
altsetting in the interface descriptors. This confuses userspace, since
alsa-lib uses device 0 by default. So lets parse interfaces in two steps:
1. Parse altsettings with PCM formats.
2. Parse altsettings with non-PCM formats.
This fixes at least following cards:
- Audinst HUD-mx2
- Audinst HUD-mini
[ Adapted to 5.3 kernel by tiwai ]
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many open code for releasing audioformat object.
Provide a unified helper and call it from the all places.
Only a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
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Merge tag 'asoc-fix-v5.3-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.3
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
As a result to support AMDTP domain, no drivers call kernel APIs to
start/stop each AMDTP stream. This commit localize these APIs.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to support AMDTP domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a kernel API to start a couple of isochronous contexts
for some AMDTP streams.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a kernel API to insert AMDTP stream to list in AMDTP
domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a kernel API to stop a couple of isochronous contexts
for AMDTP streams. The API is not protected with any lock primitive.
Callers should use this with enough lock against concurrent access.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds 'struct amdtp_domain' structure. This structure
has list of instance of AMDTP stream to handle a couple of
isochronous contexts.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
This patch fixes page faults when IOMMU is enabled.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD platform device acp3x_rv_i2s created by parent PCI device
driver. Pass struct device of the parent to
snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use
correct dma_ops. Otherwise, it will use default dma_ops which
is nommu_dma_ops on x86_64 even when IOMMU is enabled and
set to non passthrough mode.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the new PCI ID 0x1d17 0x3288 Zhaoxin controller support
Signed-off-by: Tony W Wang-oc <TonyWWang-oc@zhaoxin.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer bptr is being assigned a value that is never read
and it is being updated in the next statement with a new value.
The initialization is redundant and can be removed.
Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy HD-Audio driver cannot handle Skylake+ platforms with
digital microphones. For those platforms, the SOF or SST drivers need
to be used.
This patch provides an automatic way of detecting the presence of
DMICs using NHTL information reported by the BIOS. A kernel kconfig
option or a kernel module parameter provide an opt-in means of
stopping the probe. The kernel would then look for an alternate driver
registered for the same PCI ID to probe.
With this capability, distros no longer have to blacklist
snd-hda-intel, but still need to make sure the SOF/SST drivers are
functional by providing the relevant firmware and topology files in
/lib/firmware/intel
The coexistence between SOF and SST drivers and their dynamic
detection is not addressed by this patch, different mechanisms need to
be used, e.g. DMI-based quirks.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No functionality change, only use common functions now.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The NHLT spec defines a VENDOR_DEFINED geometry, which requires
reading additional information to figure out the number of
microphones.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move parts of the code outside of the Skylake driver to help detect
the presence of DMICs (which are not supported by the HDaudio legacy
driver).
No functionality change (except for the removal of useless OR
operations), only indentation and checkpatch fixes, making sure
that the code compiles without ACPI and fixing an ACPI leak
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Prepare move from NHLT code to common directory, starting with header.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
88200 and 96000 sampling rate was not enabled on driver, so can't be played.
The error information:
max98373 3-0031：rate 96000 not supported
max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22
Signed-off-by: fengchunguo <chunguo.feng@amlogic.com>
Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The A64 audio codec uses the original I2S block but the SR and
WSS computation currently assigned is for the newer block.
Fixes: 619c15f7fa (ASoC: sun4i-i2s: Change SR and WSS computation)
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
syzbot found the following crash on:
general protection fault: 0000 [#1] SMP KASAN
RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75
Call Trace:
snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007
snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline]
snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280
usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576
usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361
really_probe+0x281/0x650 drivers/base/dd.c:548
....
It was introduced in commit 801ebf1043 for checking pipe and endpoint
types. It is fixed by adding a check of the ep pointer in question.
BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2
Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com>
Fixes: 801ebf1043 ("ALSA: usb-audio: Sanity checks for each pipe and EP types")
Cc: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Hillf Danton <hdanton@sina.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark switch cases where we are expecting to fall through.
This patch fixes the following warnings (Building: sparc64):
sound/sparc/dbri.c: In function ‘reverse_bytes’:
sound/sparc/dbri.c:582:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:583:2: note: here
case 16:
^~~~
sound/sparc/dbri.c:584:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:585:2: note: here
case 8:
^~~~
sound/sparc/dbri.c:586:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:587:2: note: here
case 4:
^~~~
sound/sparc/dbri.c:588:5: warning: this statement may fall through [-Wimplicit-fallthrough=]
b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/sparc/dbri.c:589:2: note: here
case 2:
^~~~
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark switch cases where we are expecting to fall through.
This patch fixes the following warning (Building: m68k):
sound/oss/dmasound/dmasound_atari.c: warning: this statement may fall
through [-Wimplicit-fallthrough=]: => 1449:24
Notice that, in this particular case, the code comment is
modified in accordance with what GCC is expecting to find.
Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Tested-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Reviewed-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
lost wakeup can occur after enabling irq, therefore put task
into interruptible before enabling interrupts,
without this change, task can be put to sleep and snd_pcm_drain
will delay
Fixes: f2b3614cef ("ALSA: PCM - Don't check DMA time-out too shortly")
Signed-off-by: Yuki Tsunashima <ytsunashima@jp.adit-jv.com>
Signed-off-by: Suresh Udipi <sudipi@jp.adit-jv.com>
[ported from 4.9]
Signed-off-by: Adam Miartus <amiartus@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot of places in the driver use onyx_read_register() without
checking the return value, and it's been working OK for ~10 years
or so, so probably never fails ... Rather than trying to check the
return value everywhere, which would be relatively intrusive, at
least make sure we don't use an uninitialized value.
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2
audio interfaces. Although the interfaces are USB compliant,
additional input/output level controls and hardware routing/mixing
functionality are available using proprietary USB requests.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Distribution installation images such as Debian include different sets
of modules which can be downloaded dynamically. Such images may notably
include the hda sound modules but not the i915 DRM module, even if the
latter was enabled at build time, as reported on
https://bugs.debian.org/931507
In such a case hdac_i915 would be linked in and try to load the i915
module, fail since it is not there, but still wait for a whole minute
before giving up binding with it.
This fixes such as case by only waiting for the binding if the module
was properly loaded (or module support is disabled, in which case i915
is already compiled-in anyway).
Fixes: f9b54e1961 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Signed-off-by: Samuel Thibault <samuel.thibault@ens-lyon.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In build_adc_controls(), there is an if statement on line 773 to check
whether ak->adc_info is NULL:
if (! ak->adc_info ||
! ak->adc_info[mixer_ch].switch_name)
When ak->adc_info is NULL, it is used on line 792:
knew.name = ak->adc_info[mixer_ch].selector_name;
Thus, a possible null-pointer dereference may occur.
To fix this bug, referring to lines 773 and 774, ak->adc_info
and ak->adc_info[mixer_ch].selector_name are checked before being used.
This bug is found by a static analysis tool STCheck written by us.
Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last fallback of CORB/RIRB communication error recovery is to turn
on the single command mode, and this last resort usually means that
something is really screwed up. Instead of a normal dev_err(), show
the error more clearly with dev_WARN() with the caller stack trace.
Also, show the bus-reset fallback also as an error, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The slot_width is a property for the bus while the constraint for
SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format.
Applying slot_width constraint to sample_bits works most of the time, but
it will blacklist valid formats in some cases.
With slot_width 24 we can support S24_3LE and S24_LE formats as they both
look the same on the bus, but a a 24 constraint on sample_bits would not
allow S24_LE as it is stored in 32bits in memory.
Implement a simple hw_rule function to allow all formats which require less
or equal number of bits on the bus as slot_width (if configured).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit db51707b9c.
Revert "ASoC: rockchip: i2s: Support mono capture"
Previous discussion in
https://patchwork.kernel.org/patch/10147153/
explains the issue of the patch.
While device is configured as 1-ch, hardware is still
generating a 2-ch stream.
When user space reads the data and assumes it is a 1-ch stream,
the rate will be slower by 2x.
Revert the change so 1-ch is not supported.
User space can selectively take one channel data out of two channel
if 1-ch is preferred.
Currently, both channels record identical data.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix some typo to have the filaname given in a comment match the real name
of the file.
Some 'acpi' have erroneously been written 'apci'
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
When running McASP as master capture alone will not record any audio unless
a parallel playback stream is running. As soon as the playback stops the
captured data is going to be silent again.
In McASP master mode we need to set the PDIR for the clock pins and fix
the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above
AMUTE.
This went unnoticed as most of the boards uses McASP as slave and neither
of these issues are visible (audible) in those setups.
Fixes: ca3d943334 ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of using kmalloc + memcpy, use kmemdup
to simplify the code.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This conexant codec isn't in the supported codec list yet, the hda
generic driver can drive this codec well, but on a Lenovo machine
with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI
to make the leds work. After adding this codec to the list, the
driver patch_conexant.c will apply THINKPAD_ACPI to this machine.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the recent Intel HD-audio controller chips show a
significant stall during the system PM resume intermittently. It
doesn't happen so often and usually it may read back successfully
after one or more seconds, but in some rare worst cases the driver
went into fallback mode.
After trial-and-error, we found out that the communication stall seems
covered by issuing the sync after each verb write, as already done for
AMD and other chipsets. So this patch enables the write-sync flag for
the recent Intel chips, Skylake and onward, as a workaround.
Also, since Broxton and co have the very same driver flags as Skylake,
refer to the Skylake driver flags instead of defining the same
contents again for simplification.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901
Reported-and-tested-by: Todd Brandt <todd.e.brandt@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the DAI format setup fails, there is no valid communication format
between CPU and CODEC, so fail card instantiation, rather than continue
with a card that will most likely not function properly.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sizeof (byte) is always 1, use that fact and make interesting code explicit.
Signed-off-by: Pavel Machek <pavel@denx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
put_device will call ac97_codec_release to free
ac97_codec_device and other resources, so remove the kfree
and other redundant code.
Fixes: 74426fbff6 ("ALSA: ac97: add an ac97 bus")
Signed-off-by: Ding Xiang <dingxiang@cmss.chinamobile.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Downgrade "nothing to do in IRQ thread" message from error to a debug
message in the IPC interrupt handler thread.
The spurious wake-up can happen if a HDA stream interrupt is
raised while the IPC interrupt thread is running. IPC functionality
is not impacted by this condition, so debug is a more appropriate
trace level.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
apq8016_sbc_parse_of() sets up multiple DAI links, depending on the
number of nodes in the device tree. However, at the moment
CPU and platform components are only allocated for the first link.
This causes an oops when more than one link is defined:
Internal error: Oops: 96000044 [#1] SMP
CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4
Call trace:
apq8016_sbc_platform_probe+0x1a8/0x3f0
platform_drv_probe+0x50/0xa0
...
Move the allocation inside the loop to ensure that each link is
properly initialized.
Fixes: 98b232ca9e ("ASoC: qcom: apq8016_sbc: use modern dai_link style")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Draining makes little sense in the situation of hardware overrun, as the
hardware will have consumed all its available samples. Additionally,
draining whilst the stream is paused would presumably get stuck as no
data is being consumed on the DSP side.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Partial drain and next track are intended for gapless playback and
don't really have an obvious interpretation for a capture stream, so
makes sense to not allow those operations on capture streams.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call
snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which
allow a transition to SNDRV_PCM_STATE_SETUP. The stream should
only be able to move to the setup state once it has received a
SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing
those ioctls whilst in the open state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.
To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.
Fixes: 4f2ab5e1d1 ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in
dapm_cnew_widget() to hold the new dapm widget. Then, different actions are
taken according to the id of the widget, i.e., 'w->id'. If any failure
occurs during this process, snd_soc_dapm_new_control_unlocked() should be
terminated by going to the 'request_failed' label. However, the allocated
kernel buffer is not freed on this code path, leading to a memory leak bug.
To fix the above issue, free the buffer before returning from
snd_soc_dapm_new_control_unlocked() through the 'request_failed' label.
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
MOTU data block processing layer has some tracepoints events. This
commit is code refactoring to split probing the events from processing
data blocks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes each of data block processing layer so that it
receives list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for FF data block processing layer so that
it can receive list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for MOTU data block processing layer so that
it can receive list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for TASCAM data block processing layer so that
it can receive list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for DOT data block processing layer so that
it can receive list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for AM824 data block processing layer so that
it can receive list of packet descriptor.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes signature of callback function to call data block
processing layer with packet descriptor. At present, the layer is called
per packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring for common processing for payloads of
ishocornous context.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is code refactoring to separate PCM substream processing from packet
queueing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit uses packet descriptor to parse headers of IR context and
parse timing information for sequence transferred by target device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit uses packet descriptor to parse headers of IT context and
generate timing information for ideal sequence.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation of ALSA IEC 61883-1/6 packet streaming engine,
16 packets are handled in one interrupt of isochronous context of OHCI
1394.
Overall packet processing runs for each. However, this is not better in
a point to split the processing into several parts.
This commit is an attempt to add intermediate representation for
parameters required for the parts.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a previous commit, the variable passed from packet streaming layer
for syt variable is useless. This commit obsoletes it.
In my future work, the syt information is passed to data block processing
layer by another way.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some protocols don't use syt field of CIP header to represent
presentation timestamp. For such protocol, ALSA IEC 61883-1/6
packet streaming engine uses a pointer into local variable for
the value of syt to call data block processing layer. However,
it can decide the value when initializing packet streaming
layer.
This commit adds 'syt_override' member for packet streaming
layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA IEC 61883-1/6 packet streaming engine, two types of data block
counter are supported. This commit applies code refactoring to make it
explicitly for IR context.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA IEC 61883-1/6 packet streaming engine, two types of data block
counter are supported. This commit applies code refactoring to make it
explicitly for IT context.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preparation for future commit that 'struct
amdtp_stream.data_block_count' does not represent the value of
data block count for current data block.
However, data block count is required for calculation of sequence index
in sequence-multiplied data channel. Some of data block processing layer
require it; e.g. for AM824 data block.
This commit passes data block count to the processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preparation for future commit that 'struct
amdtp_stream.data_block_count' does not represent the value of
data block count for current data block.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recent firmware for Fireworks board module have a quirk to start
transmission of CIP with non-zero value for its data block counter.
In current implementation of ALSA firewire stack, the quirk is handled
by 'struct amdtp_stream.ctx_data.tx.first_dbc' with value 0x02. However,
the value comes from reverse engineering. It's better to handle this
quirk without the explicit value.
In a process to parse CIP header, the quirk of data block counter
affects decision of sequence index in sequence-multiplexed data channel;
i.e. MIDI conformant data channel. In Fireworks, the index is decided
by the number of data blocks from top of the same CIP, thus the value
of data block counter is useless.
This commit adds CIP_UNALIGHED_DBC flag and obsoletes the explicit
value for this quirk.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TS3A227E says that the headset keypress detection needs the MICBIAS
power in order to report the key events to ensure proper operation
The headset keypress detection needs the MICBIAS power in order to report
the key events all the time as long as MIC is present. So MICBIAS pin
is forced on when a MICROPHONE is detected.
On Veyron Minnie I observed that if the MICBIAS power is not present and
the key press detection is activated (just because it is enabled when you
insert a headset), it randomly reports a keypress on insert.
E.g. (KEY_PLAYPAUSE)
Event: (SW_HEADPHONE_INSERT), value 1
Event: (SW_MICROPHONE_INSERT), value 1
Event: -------------- SYN_REPORT ------------
Event: (KEY_PLAYPAUSE), value 1
Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying
effect that the media player starts a play/pause loop.
Note that, although most of the time the key reported is the one
associated with BTN_0, not always this is true. On my tests I also saw
different keys reported
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20190719173929.24065-1-enric.balletbo@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When sample rate of TX is different with sample rate of RX in
async mode, the MFreq selection will be wrong.
For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz.
Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX
and RX instance, the correct value of MFreq is 2 for both TX and RX.
But original method will cause MFreq = 0 for TX, MFreq = 2 for RX.
If TX is started after RX, RX will be impacted, RX work abnormal with
MFreq = 0.
This patch is to select proper MFreq value according to TX rate and
RX rate.
Fixes: 0c516b4ff8 ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/20190716094547.46787-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a
list of the widgets connected to a specific front end DAI so it
can search through this list for available back end DAIs. The
custom_stop_condition was added to is_connected_ep to facilitate this
list not containing more widgets than is necessary. Doing so both
speeds up the DPCM handling as less widgets need to be searched and
avoids issues with CODEC to CODEC links as these would be confused
with back end DAIs if they appeared in the list of available widgets.
custom_stop_condition was implemented by aborting the graph walk
when the condition is triggered, however there is an issue with this
approach. Whilst walking the graph is_connected_ep should update the
endpoints cache on each widget, if the walk is aborted the number
of attached end points is unknown for that sub-graph. When the stop
condition triggered, the original patch ignored the triggering widget
and returned zero connected end points; a later patch updated this
to set the triggering widget's cache to 1 and return that. Both of
these approaches result in inaccurate values being stored in various
end point caches as the values propagate back through the graph,
which can result in later issues with widgets powering/not powering
unexpectedly.
As the original goal was to reduce the size of the widget list passed
to the DPCM code, the simplest solution is to limit the functionality
of the custom_stop_condition to the widget list. This means the rest
of the graph will still be processed resulting in correct end point
caches, but only widgets up to the stop condition will be added to the
returned widget list.
Fixes: 6742064aef ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets")
Fixes: 5fdd022c20 ("ASoC: dpcm: play nice with CODEC<->CODEC links")
Fixes: 09464974ea ("ASoC: dapm: Fix to return correct path list in is_connected_ep.")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD/ATI and Nvidia HDMI codec drivers didn't have the audio component
binding like i915, but it worked only with the traditional HD-audio
unsolicited event for the HDMI hotplug detection and the ELD read-up
thereafter. This has been a problem in many ways: first of all, it
goes through the hardware event transition (from GPU register write,
HD-audio controller trigger, and finally to HD-audio unsolicited event
handling), which is often unreliable and may miss some opportunities.
Second, each unsol event handling and ELD read-up need the explicit
power up / down when the codec is in the runtime suspend. Last but
not least, which is the most important, the hotplug wakeup may be
missed when the HD-audio controller is in runtime suspend. Especially
the last point is a big problem due to the recent change relevant with
vga_switcheroo that forcibly enables the runtime PM for AMD HDMI
controllers.
These issues are solved by introducing the audio component; the
hotplug notification is done by a direct function callback, which is
more accurate and reliable, and it can be processed without the actual
hardware access, i.e. no runtime PM trigger is needed, and the
HD-audio gets the event even if it's in runtime suspend. The same for
ELD query, as it's read directly from the cached ELD bytes stored in
the DRM driver, hence the whole hardware access can be skipped.
So here it is: this patch implements the audio component binding with
AMD/ATI and Nouveau DRM drivers. The biggest difference from i915
implementation is that this binding is fully optional and it can be
enabled asynchronously on the fly. That is, the driver will switch
from the HD-audio unsolicited event to the notify callback once when
the DRM component gets bound. Similarly, when DRM driver gets
unloaded, the HDMI event handling returns to the legacy mode, too.
Also, another difference from i915 is that the new code registers the
component in the codec driver, while i915 HDMI codec assumes the
component binding was already done in the HD-audio controller driver.
Hence the new code does need to de-register the component binding at
the codec exit, too.
Some other details:
- The match component ops assumes that both VGA and HD-audio
controller PCI entries belong to the same PCI bus, and only accepts
such an entry.
- The pin2port audio_ops is implemented with assumption of the fixed
widget layout. For AMD, it's starting from 3, with step 2 (3, 5, 7,
...), while for Nvidia, it's starting from 4, with step 1 (4, 5, 6,
...)
As of this patch, the corresponding component isn't implemented in DRM
side, so this change alone won't give any benefit. By the following
changes in DRM sides, the mission will be completed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDMI jack handling reports the state change always via
snd_jack_report() whenever hdmi_present_sense() is called, even if the
state itself doesn't change from the previous time. This is mostly
harmless but still a bit confusing to user-space.
This patch reduces such spurious jack state changes and reports only
when the state really changed. Also, as a minor optimization, avoid
overwriting the pin ELD data when the state is identical.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent rewrite of PCM link lock management introduced the refcount
in snd_pcm_group object, managed by the kernel refcount_t API. This
caused unexpected kernel warnings when the kernel is built with
CONFIG_REFCOUNT_FULL=y. As the warning line indicates, the problem is
obviously that we start with refcount=0 and do refcount_inc() for
adding each PCM link, while refcount_t API doesn't like refcount_inc()
performed on zero.
For adapting the proper refcount_t usage, this patch changes the logic
slightly:
- The initial refcount is 1, assuming the single list entry
- The refcount is incremented / decremented at each PCM link addition
and deletion
- ... which allows us concentrating only on the refcount as a release
condition
Fixes: f57f3df03a ("ALSA: pcm: More fine-grained PCM link locking")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204221
Reported-and-tested-by: Duncan Overbruck <kernel@duncano.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes.
- The optimization of PM resume with HD-audio HDMI codecs, which
eventually work around weird issues
- A correction of Intel Icelake HDMI audio code
- Quirks for Dell machines with Realtek HD-audio codecs
- The fix for too long sequencer write stall that was spotted by
syzkaller
- A few trivial cleanups reported by coccinelle
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Merge tag 'sound-fix-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes.
- The optimization of PM resume with HD-audio HDMI codecs, which
eventually work around weird issues
- A correction of Intel Icelake HDMI audio code
- Quirks for Dell machines with Realtek HD-audio codecs
- The fix for too long sequencer write stall that was spotted by
syzkaller
- A few trivial cleanups reported by coccinelle"
* tag 'sound-fix-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Don't resume forcibly i915 HDMI/DP codec
ALSA: hda/hdmi - Fix i915 reverse port/pin mapping
ALSA: hda/hdmi - Remove duplicated define
ALSA: seq: Break too long mutex context in the write loop
ALSA: hda/realtek: apply ALC891 headset fixup to one Dell machine
ALSA: rme9652: Unneeded variable: "result".
ALSA: emu10k1: Remove unneeded variable "change"
ALSA: au88x0: Remove unneeded variable: "changed"
ALSA: hda/realtek - Fixed Headphone Mic can't record on Dell platform
ALSA: ps3: Remove Unneeded variable: "ret"
ALSA: lx6464es: Remove unneeded variable err
Commit 7b9584fa1c ("staging: line6: Move altsetting to properties")
set a wrong altsetting for LINE6_PODHD500_1 during refactoring.
Set the correct altsetting number to fix the issue.
BugLink: https://bugs.launchpad.net/bugs/1790595
Fixes: 7b9584fa1c ("staging: line6: Move altsetting to properties")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
