Hui Wang [Wed, 4 Sep 2019 05:53:27 +0000 (13:53 +0800)]
ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentre
This ThinkCentre machine has a new realtek codec alc222, it is not
in the support list, we add it in the realtek.c then this machine
can apply FIXUPs for the realtek codec.
And this machine has two front mics which can't be handled
by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as
a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't
work on this machine. Fortunately another FIXUP
ALC283_FIXUP_HEADSET_MIC also can change the location for one of the
two mics on this machine.
1. Has duplicated internal speakers (0x14 & 0x17) which makes the output
route become confused. So, the output volume cannot be changed by
setting.
2. Misses the headset mic pin node.
This patch disables the confusing speaker (NID 0x14) and enables the
headset mic (NID 0x19).
The recent change to shuffle the codec initialization procedure for
Realtek via commit 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on
later") caused the silent output on some machines. This change was
supposed to be safe, but it isn't actually; some devices have quirk
setups to override the EAPD via COEF or BTL in the additional verb
table, which is applied at the beginning of snd_hda_gen_init(). And
this EAPD setup is again overridden in alc_auto_init_amp().
For recovering from the regression, tell snd_hda_gen_init() not to
apply the verbs there by a new flag, then apply the verbs in
alc_init().
Takashi Iwai [Thu, 29 Aug 2019 07:52:02 +0000 (09:52 +0200)]
ALSA: hda - Fix potential endless loop at applying quirks
Since the chained quirks via chained_before flag is applied before the
depth check, it may lead to the endless recursive calls, when the
chain were set up incorrectly. Fix it by moving the depth check at
the beginning of the loop.
Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Mon, 26 Aug 2019 13:55:15 +0000 (22:55 +0900)]
ALSA: oxfw: fix to handle correct stream for PCM playback
When userspace application calls ioctl(2) to configure hardware for PCM
playback substream, ALSA OXFW driver handles incoming AMDTP stream.
In this case, outgoing AMDTP stream should be handled.
This commit fixes the bug for v5.3-rc kernel.
Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sun, 25 Aug 2019 07:21:44 +0000 (09:21 +0200)]
ALSA: seq: Fix potential concurrent access to the deleted pool
The input pool of a client might be deleted via the resize ioctl, the
the access to it should be covered by the proper locks. Currently the
only missing place is the call in snd_seq_ioctl_get_client_pool(), and
this patch papers over it.
Takashi Iwai [Tue, 20 Aug 2019 19:43:42 +0000 (21:43 +0200)]
ALSA: usb-audio: Check mixer unit bitmap yet more strictly
The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a
variable size depending on both input and output pins. Its size is to
fit with input * output bits. The problem is that the input size
can't be determined simply from the unit descriptor itself but it
needs to parse the whole connected sources. Although the
uac_mixer_unit_get_channels() tries to check some possible overflow of
this bitmap, it's incomplete due to the lack of the evaluation of
input pins.
For covering possible overflows, this patch adds the bitmap overflow
check in the loop of input pins in parse_audio_mixer_unit().
Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 21 Aug 2019 18:00:02 +0000 (20:00 +0200)]
ALSA: line6: Fix memory leak at line6_init_pcm() error path
I forgot to release the allocated object at the early error path in
line6_init_pcm(). For addressing it, slightly shuffle the code so
that the PCM destructor (pcm->private_free) is assigned properly
before all error paths.
Takashi Iwai [Thu, 15 Aug 2019 09:41:06 +0000 (11:41 +0200)]
ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()
The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code. It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.
This patch fixes the check before the conversion.
While we're at it, correct a typo in the comment in the function,
too.
Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jeronimo Borque [Mon, 19 Aug 2019 01:35:38 +0000 (22:35 -0300)]
ALSA: hda - Fixes inverted Conexant GPIO mic mute led
"enabled" parameter historically referred to the device input or
output, not to the led indicator. After the changes added with the led
helper functions the mic mute led logic refers to the led and not to
the mic input which caused led indicator to be negated.
Fixing logic in cxt_update_gpio_led and updated
cxt_fixup_gpio_mute_hook
Also updated debug messages to ease further debugging if necessary.
Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper") Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hui Peng [Thu, 15 Aug 2019 04:31:34 +0000 (00:31 -0400)]
ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_term
`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.
This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).
Hui Peng [Wed, 14 Aug 2019 02:34:04 +0000 (22:34 -0400)]
ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unit
The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.
Hui Wang [Wed, 14 Aug 2019 04:09:08 +0000 (12:09 +0800)]
ALSA: hda - Add a generic reboot_notify
Make codec enter D3 before rebooting or poweroff can fix the noise
issue on some laptops. And in theory it is harmless for all codecs
to enter D3 before rebooting or poweroff, let us add a generic
reboot_notify, then realtek and conexant drivers can call this
function.
Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hui Wang [Wed, 14 Aug 2019 04:09:07 +0000 (12:09 +0800)]
ALSA: hda - Let all conexant codec enter D3 when rebooting
We have 3 new lenovo laptops which have conexant codec 0x14f11f86,
these 3 laptops also have the noise issue when rebooting, after
letting the codec enter D3 before rebooting or poweroff, the noise
disappers.
Instead of adding a new ID again in the reboot_notify(), let us make
this function apply to all conexant codec. In theory make codec enter
D3 before rebooting or poweroff is harmless, and I tested this change
on a couple of other Lenovo laptops which have different conexant
codecs, there is no side effect so far.
Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Wenwen Wang [Sat, 10 Aug 2019 04:29:48 +0000 (23:29 -0500)]
ALSA: hda - Fix a memory leak bug
In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc().
Then, the pin widgets in 'codec' are parsed. However, if the parsing
process fails, 'spec' is not deallocated, leading to a memory leak.
To fix the above issue, free 'spec' before returning the error.
Takashi Iwai [Fri, 9 Aug 2019 09:23:00 +0000 (11:23 +0200)]
ALSA: hda - Apply workaround for another AMD chip 1022:1487
MSI MPG X570 board is with another AMD HD-audio controller (PCI ID
1022:1487) and it requires the same workaround applied for X370, etc
(PCI ID 1022:1457).
Wenwen Wang [Thu, 8 Aug 2019 05:50:58 +0000 (00:50 -0500)]
ALSA: firewire: fix a memory leak bug
In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.
To fix the above issue, free 'b->packets' before returning the error code.
Wenwen Wang [Thu, 8 Aug 2019 05:15:21 +0000 (00:15 -0500)]
sound: fix a memory leak bug
In sound_insert_unit(), the controlling structure 's' is allocated through
kmalloc(). Then it is added to the sound driver list by invoking
__sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is
removed from the list through __sound_remove_unit(). If 'index' is not less
than 0, -EBUSY is returned to indicate the error. However, 's' is not
deallocated on this execution path, leading to a memory leak bug.
To fix the above issue, free 's' before -EBUSY is returned.
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 6 Aug 2019 15:31:48 +0000 (17:31 +0200)]
ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
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.
Wenwen Wang [Wed, 7 Aug 2019 09:08:51 +0000 (04:08 -0500)]
ALSA: hiface: fix multiple memory leak bugs
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'.
Takashi Iwai [Tue, 6 Aug 2019 12:03:56 +0000 (14:03 +0200)]
ALSA: hda - Don't override global PCM hw info flag
The commit bfcba288b97f ("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: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Wenwen Wang [Tue, 6 Aug 2019 07:00:27 +0000 (03:00 -0400)]
ALSA: usb-audio: fix a memory leak bug
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: 7edf3b5e6a45 ("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>
Takashi Iwai [Tue, 6 Aug 2019 10:28:08 +0000 (12:28 +0200)]
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.
ASoC: amd: acp3x: use dma address for acp3x dma driver
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.
ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
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.
Samuel Thibault [Fri, 26 Jul 2019 21:47:02 +0000 (23:47 +0200)]
ALSA: hda: Fix 1-minute detection delay when i915 module is not available
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).
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).
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.
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.
Hui Wang [Thu, 25 Jul 2019 06:57:37 +0000 (14:57 +0800)]
ALSA: hda - Add a conexant codec entry to let mute led work
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>
ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips
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.
ASoC: Fail card instantiation if DAI format setup fails
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.
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.
Stephan Gerhold [Mon, 22 Jul 2019 13:03:52 +0000 (15:03 +0200)]
ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
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:
Move the allocation inside the loop to ensure that each link is
properly initialized.
Fixes: 98b232ca9e0e ("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>
Charles Keepax [Mon, 22 Jul 2019 09:24:36 +0000 (10:24 +0100)]
ALSA: compress: Be more restrictive about when a drain is allowed
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.
Charles Keepax [Mon, 22 Jul 2019 09:24:35 +0000 (10:24 +0100)]
ALSA: compress: Don't allow paritial drain operations on capture streams
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.
Charles Keepax [Mon, 22 Jul 2019 09:24:34 +0000 (10:24 +0100)]
ALSA: compress: Prevent bypasses of set_params
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.
Charles Keepax [Mon, 22 Jul 2019 09:24:33 +0000 (10:24 +0100)]
ALSA: compress: Fix regression on compressed capture streams
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.
Wenwen Wang [Mon, 22 Jul 2019 13:57:44 +0000 (08:57 -0500)]
ASoC: dapm: fix a memory leak bug
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.
ASoC: SOF: use __u32 instead of uint32_t in uapi headers
When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to
make sure they can be included from user-space.
Currently, header.h and fw.h are excluded from the test coverage.
To make them join the compile-test, we need to fix the build errors
attached below.
For a case like this, we decided to use __u{8,16,32,64} variable types
in this discussion:
https://lkml.org/lkml/2019/6/5/18
Build log:
CC usr/include/sound/sof/header.h.s
CC usr/include/sound/sof/fw.h.s
In file included from <command-line>:32:0:
./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’
uint32_t magic; /**< 'S', 'O', 'F', '\0' */
^~~~~~~~
./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’
uint32_t type; /**< component specific type */
^~~~~~~~
./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’
uint32_t size; /**< size in bytes of data excl. this struct */
^~~~~~~~
./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’
uint32_t abi; /**< SOF ABI version */
^~~~~~~~
./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’
uint32_t reserved[4]; /**< reserved for future use */
^~~~~~~~
./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’
uint32_t data[0]; /**< Component data - opaque to core */
^~~~~~~~
In file included from <command-line>:32:0:
./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’
uint32_t size; /* bytes minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’
uint32_t offset; /* offset from base */
^~~~~~~~
./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’
uint32_t size; /* bytes minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’
uint32_t num_blocks; /* number of blocks */
^~~~~~~~
./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’
uint32_t file_size; /* size of file minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’
uint32_t num_modules; /* number of modules */
^~~~~~~~
./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’
uint32_t abi; /* version of header format */
^~~~~~~~
SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection
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
Shengjiu Wang [Tue, 16 Jul 2019 09:45:47 +0000 (17:45 +0800)]
ASoC: cs42xx8: Fix MFREQ selection issue for async mode
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.
Charles Keepax [Thu, 18 Jul 2019 08:43:33 +0000 (09:43 +0100)]
ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks
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: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("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>
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
ALSA: hda - Optimize resume for codecs without jack detection
The codecs without jack detection also don't have to be resumed
forcibly because, obviously, they have no jack. Skip the forced
resume in such a case as optimization as well.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
We apply the codec resume forcibly at system resume callback for
updating and syncing the jack detection state that may have changed
during sleeping. This is, however, superfluous for the codec like
Intel HDMI/DP, where the jack detection is managed via the audio
component notification; i.e. the jack state change shall be reported
sooner or later from the graphics side at mode change.
This patch changes the codec resume callback to avoid the forcible
resume conditionally with a new flag, codec->relaxed_resume, for
reducing the resume time. The flag is set in the codec probe.
Although this doesn't fix the entire bug mentioned in the bugzilla
entry below, it's still a good optimization and some improvements are
seen.
Wen Yang [Sat, 13 Jul 2019 03:46:14 +0000 (11:46 +0800)]
ASoC: samsung: odroid: fix an use-after-free issue for codec
The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.
Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI") Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Cc: Krzysztof Kozlowski <krzk@kernel.org> Cc: Sangbeom Kim <sbkim73@samsung.com> Cc: Sylwester Nawrocki <s.nawrocki@samsung.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for Icelake HDMI codec introduced the mapping from pin
NID to the i915 gfx port number. However, it forgot the reverse
mapping from the port number to the pin NID that is used in the ELD
notifier callback. As a result, it's processed to a wrong widget and
gives a warning like
snd_hda_codec_hdmi hdaudioC0D2: HDMI: pin nid 5 not registered
This patch corrects it with a proper reverse mapping function.
ALSA: seq: Break too long mutex context in the write loop
The fix for the racy writes and ioctls to sequencer widened the
application of client->ioctl_mutex to the whole write loop. Although
it does unlock/relock for the lengthy operation like the event dup,
the loop keeps the ioctl_mutex for the whole time in other
situations. This may take quite long time if the user-space would
give a huge buffer, and this is a likely cause of some weird behavior
spotted by syzcaller fuzzer.
This patch puts a simple workaround, just adding a mutex break in the
loop when a large number of events have been processed. This
shouldn't hit any performance drop because the threshold is set high
enough for usual operations.
Fixes: 7bd800915677 ("ALSA: seq: More protection for concurrent write and ioctl races") Reported-by: syzbot+97aae04ce27e39cbfca9@syzkaller.appspotmail.com Reported-by: syzbot+4c595632b98bb8ffcc66@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/au88x0/au88x0_a3d.c:821:8-15: Unneeded variable: "changed".
Return "1" on line 834
sound/pci/au88x0/au88x0_a3d.c:768:5-12: Unneeded variable: "changed".
Return "1" on line 777
sound/pci/au88x0/au88x0_a3d.c:804:5-12: Unneeded variable: "changed".
Return "1" on line 813
sound/pci/au88x0/au88x0_a3d.c:786:8-15: Unneeded variable: "changed".
Return "1" on line 796
Kailang Yang [Mon, 15 Jul 2019 02:41:50 +0000 (10:41 +0800)]
ALSA: hda/realtek - Fixed Headphone Mic can't record on Dell platform
It assigned to wrong model. So, The headphone Mic can't work.
Fixes: 3f640970a414 ("ALSA: hda - Fix headset mic detection problem for several Dell laptops") Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: audio-graph-card: add missing const at graph_get_dai_id()
commit c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in
graph_get_dai_id()") fixups use-after-free issue,
but, it need to use "const" for reg. This patch adds it.
We will have below without this patch
LINUX/sound/soc/generic/audio-graph-card.c: In function 'graph_get_dai_id':
LINUX/sound/soc/generic/audio-graph-card.c:87:7: warning: assignment discards\
'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
reg = of_get_property(node, "reg", NULL);
Fixes: c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Wen Yang <wen.yang99@zte.com.cn> Link: https://lore.kernel.org/r/87sgrd43ja.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
Shuming Fan [Thu, 11 Jul 2019 08:22:14 +0000 (16:22 +0800)]
ASoC: rt1011: fix DC calibration offset not applying
There are two issues to fix:
- DC offset calibration data will be reset after stopping playback.
- DC offset calibration data should be applied in the initial setting.
Wen Yang [Wed, 10 Jul 2019 07:25:09 +0000 (15:25 +0800)]
ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()
After calling of_node_put() on the node variable, it is still being
used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.
Fixes: a0c426fe1433 ("ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()") Link: https://lore.kernel.org/r/1562743509-30496-5-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Wen Yang [Wed, 10 Jul 2019 07:25:08 +0000 (15:25 +0800)]
ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm()
After calling of_node_put() on the ports, port, and node variables,
they are still being used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.
Fixes: dd98fbc558a0 ("ASoC: audio-graph-card: cleanup DAI link loop method - step1") Link: https://lore.kernel.org/r/1562743509-30496-4-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Wen Yang [Wed, 10 Jul 2019 07:25:07 +0000 (15:25 +0800)]
ASoC: simple-card: fix an use-after-free in simple_for_each_link()
The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.
Fixes: d947cdfd4be2 ("ASoC: simple-card: cleanup DAI link loop method - step1") Link: https://lore.kernel.org/r/1562743509-30496-3-git-send-email-wen.yang99@zte.com.cn Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: simple-card-utils: care no Platform for DPCM
commit 34614739988ad ("ASoC: soc-core: support dai_link with
platforms_num != 1") supports multi Platform, and
commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform
possibility") removed no Platform from simple-card.
Multi Platform is now checking both Platform name/of_node are NULL case.
But in normal case, DPCM be doesn't have Platform.
asoc_simple_canonicalize_platform() try to use CPU of_node
to Platform (This is needed for DMAEngine platform case),
but it still might be NULL at DPCM be.
This patch try to use no Platform after that if Platform of_node
is still NULL. It can't probe without this patch.
max98357a_daiops_trigger() is possible to be called in atomic context if
the .nonatomic flag is equal to 0 in the DAI links.
When cancel_delayed_work_sync() in max98357a_daiops_trigger() is called
in atomic context, kernel emits the following message: "BUG: sleeping
function called from invalid context".
According to the DT binding document, value less than or equal to 5ms of
sdmod-delay should be sufficient to avoid the pop noise. Use mdelay
(i.e. busy loop) for such low delay should be acceptable.
Fixes: cec5b01f8f1c ("ASoC: max98357a: avoid speaker pop when playback
startup")
ALSA: firewire-lib: code refactoring for post operation to data block counter
As a result of former commits, post operation to data block count for
cases without CIP_DBC_IS_END_EVENT can be done just with
data_block_counter member of amdtp_stream structure.
This commit adds code refactoring to obsolete local variable for
data block counter.
ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
For IT context, tracepoints event is probed after calculating next data
block counter. This brings difference of data block counter between
the probed event and actual isochronous packet.
ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
For IR context, ALSA IEC 61883-1/6 engine uses initial value of data
block counter as UINT_MAX, to detect first isochronous packet in the
middle of packet streaming.
At present, when CIP_DBC_IS_END_EVENT is not used (i.e. for drivers except
for ALSA fireworks driver), the initial value is used as is for
tracepoints event. However, the engine can detect the value of dbc field
in the payload of first isochronous packet and the value should be assigned
to the event.
This commit fixes the bug.
Fixes: 76864868dbab ("ALSA: firewire-lib: cache next data_block_counter after probing tracepoints event for IR context") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
For IR context, ALSA IEC 61883-1/6 engine uses initial value of data
block counter as UINT_MAX, to detect first isochronous packet in the
middle of packet streaming.
At present, when CIP_NO_HEADER is used (i.e. for ALSA fireface driver),
the initial value is used for tracepoints event. 0x00 should be
for the event when the initial value is UINT_MAX because isochronous
packets with CIP_NO_HEADER option has no field for data block count.
This commit fixes the bug.
Fixes: 76864868dbab ("ALSA: firewire-lib: cache next data_block_counter after probing tracepoints event for IR context") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
Although CIP header is handled as context header, the length of isochronous
packet includes two quadlets for its payload. In tracepoints event the
value of payload_quadlets should includes the two quadlets. But at present
it doesn't.
This commit fixes the bug.
Fixes: b18f0cfaf16b ("ALSA: firewire-lib: use 8 byte packet header for IT context to separate CIP header from CIP payload") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge tag 'asoc-v5.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.3
This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.
- Lots more work on moving towards a component based framework from
Morimoto-san.
- Support for force disconnecting muxes from Jerome Brunet.
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308.
firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA firewire-motu driver allows applications of rawmidi interface to
start packet streaming for transmission of MIDI messages. However at
error path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA firewire-digi00x driver allows applications of rawmidi interface to
start packet streaming for transmission of MIDI messages. However at
error path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA dice driver allows applications of rawmidi interface to start
packet streaming for transmission of MIDI messages. However at error
path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA oxfw driver allows applications of rawmidi interface to start
packet streaming for transmission of MIDI messages. However at error
path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
This commit fixes the bug.
Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA fireworks driver allows applications of rawmidi interface to start
packet streaming for transmission of MIDI messages. However at error
path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
This commit fixes the bug.
Fixes: 3d7250667ea9 ("ALSA: fireworks: configure sampling transfer frequency in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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