More fixes for v5.11, almost all driver specific issues including new
device IDs - there's one error handling fix for the topology stuff too.
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Merge tag 'asoc-fix-v5.11-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.11
More fixes for v5.11, almost all driver specific issues including new
device IDs - there's one error handling fix for the topology stuff too.
We've had several reports of broken dependencies. The 'right' fix is
to revisit the module dependencies as suggested by Arnd Bergmann. This
is WIP at https://github.com/thesofproject/linux/pull/2683. Since this
is taking longer than expected, I am only sharing quick fixes for now.
Pierre-Louis Bossart (2):
ASoC: SOF: Intel: soundwire: fix select/depend unmet dependencies
ASoC: SOF: SND_INTEL_DSP_CONFIG dependency
sound/soc/sof/intel/Kconfig | 3 ++-
sound/soc/sof/sof-acpi-dev.c | 11 ++++++-----
sound/soc/sof/sof-pci-dev.c | 10 ++++++----
3 files changed, 14 insertions(+), 10 deletions(-)
--
2.25.1
Add flag "SOF_RT711_JD_SRC_JD2", flag "SOF_RT715_DAI_ID_FIX"
and "SOF_SDW_FOUR_SPK" to the Dell TGL-H based SKU "0A5E".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Co-developed-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210125081117.814488-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "dai_id" given into LPAIF_INTFDMA_REG(...) is already the real
DAI ID, not an index into v->dai_driver. Looking it up again seems
entirely redundant.
For IPQ806x (and SC7180 since commit 09a4f6f5d2
("ASoC: dt-bindings: lpass: Fix and common up lpass dai ids") this is
now often an out-of-bounds read because the indexes in the "dai_driver"
array no longer match the actual DAI ID.
Cc: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210125104442.135899-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
MT8192 determines the I2S clock rates according to the sampling rates.
There is only 1 set of I2S in between MT8192 and RT5682. If playing and
capturing via RT5682 in different sampling rates, the I2S data will be
corrupted.
Adds format constraints to the corresponding DAI links to make sure the
sampling rates are symmetric.
Fixes: 18b13ff23f ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210125061453.1056535-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reset (aka power off) happens when the reset gpio is made active.
Change function name to ak4458_reset to match devicetree property "reset-gpios"
Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Reviewed-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/ce650f47-4ff6-e486-7846-cc3d033f3601@blennerhassett.gen.nz
Signed-off-by: Mark Brown <broonie@kernel.org>
The sof-pci-dev driver fails to link when built into the kernel
and CONFIG_SND_INTEL_DSP_CONFIG is set to =m:
arm-linux-gnueabi-ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe':
sof-pci-dev.c:(.text+0x1c): undefined reference to `snd_intel_dsp_driver_probe'
As a temporary fix, use IS_REACHABLE to prevent the problem from
happening. A more complete solution is to move this code to
Intel-specific parts, restructure the drivers and Kconfig as discussed
with Arnd Bergmann and Takashi Iwai.
Fixes: 82d9d54a6c ("ALSA: hda: add Intel DSP configuration / probe code")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210122005725.94163-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The LKP bot reports the following issue:
WARNING: unmet direct dependencies detected for SOUNDWIRE_INTEL
Depends on [m]: SOUNDWIRE [=m] && ACPI [=y] && SND_SOC [=y]
Selected by [y]:
- SND_SOC_SOF_INTEL_SOUNDWIRE [=y] && SOUND [=y] && !UML &&
SND [=y] && SND_SOC [=y] && SND_SOC_SOF_TOPLEVEL [=y] &&
SND_SOC_SOF_INTEL_TOPLEVEL [=y] && SND_SOC_SOF_INTEL_PCI [=y]
This comes from having tristates being configured independently, when
in practice the CONFIG_SOUNDWIRE needs to be aligned with the SOF
choices: when the SOF code is compiled as built-in, the
CONFIG_SOUNDWIRE also needs to be 'y'.
The easiest fix is to replace the 'depends' with a 'select' and have a
single user selection to activate SoundWire on Intel platforms. This
still allows regmap to be compiled independently as a module.
This is just a temporary fix, the select/depend usage will be
revisited and the SOF Kconfig re-organized, as suggested by Arnd
Bergman.
Reported-by: kernel test robot <lkp@intel.com>
Fixes: a115ab9b8b ('ASoC: SOF: Intel: add build support for SoundWire')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210122005725.94163-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding PCI id for TGL-H. Like for other TGL platforms, SOF is used if
Soundwire codecs or PCH-DMIC is detected.
Signed-off-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Xiuli Pan <xiuli.pan@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210125083051.828205-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced sample rate validation code seems causing a
problem on some devices; namely, after performing this, the bus gets
screwed and it influences even on other USB devices.
As a quick workaround, perform it only for the necessary devices;
currently MOTU devices are known to need the valid altset checks, so
filter out other devices.
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Reported-by: Jamie Heilman <jamie@audible.transient.net>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203
Link: https://lore.kernel.org/r/20210123155842.22652-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix for a long-standing USB-audio bug required one more dependency
variable to be added to the hw constraints. Unfortunately I didn't
realize at debugging that the new addition may result in the overflow
of the dependency array of each snd_pcm_hw_rule (up to three plus a
sentinel), because USB-audio driver adds one more dependency only for
a certain device and bus, hence it works as is for many devices. But
in a bad case, a simple open always results in -EINVAL (with kernel
WARNING if CONFIG_SND_DEBUG is set) no matter what is passed.
Since the dependencies are real and unavoidable (USB-audio restricts
the hw_params per looping over the format/rate/channels combos), the
only good solution seems to raise the bar for one more dependency for
snd_pcm_hw_rule -- so does this patch: now the hw constraint
dependencies can be up to four.
Fixes: 506c203cc3 ("ALSA: usb-audio: Fix hw constraints dependencies")
Reported-by: Jamie Heilman <jamie@audible.transient.net>
Link: https://lore.kernel.org/r/20210123155730.22576-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clear struct snd_ctl_elem_value before calling ->put() to avoid any data
leak.
Signed-off-by: Ricardo Ribalda <ribalda@chromium.org>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://lore.kernel.org/r/20210121171644.131059-2-ribalda@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Suspending/resuming with an HDMI dongle attached leads to crashes from
an audio regmap.
Unable to handle kernel paging request at virtual address ffffffc018068000
Mem abort info:
ESR = 0x96000047
EC = 0x25: DABT (current EL), IL = 32 bits
SET = 0, FnV = 0
EA = 0, S1PTW = 0
Data abort info:
ISV = 0, ISS = 0x00000047
CM = 0, WnR = 1
swapper pgtable: 4k pages, 39-bit VAs, pgdp=0000000081b12000
[ffffffc018068000] pgd=0000000275d14003, pud=0000000275d14003, pmd=000000026365d003, pte=0000000000000000
Internal error: Oops: 96000047 [#1] PREEMPT SMP
Call trace:
regmap_mmio_write32le+0x2c/0x40
regmap_mmio_write+0x48/0x6c
_regmap_bus_reg_write+0x34/0x44
_regmap_write+0x100/0x150
regcache_default_sync+0xc0/0x138
regcache_sync+0x188/0x26c
lpass_platform_pcmops_resume+0x48/0x54 [snd_soc_lpass_platform]
snd_soc_component_resume+0x28/0x40
soc_resume_deferred+0x6c/0x178
process_one_work+0x208/0x3c8
worker_thread+0x23c/0x3e8
kthread+0x144/0x178
ret_from_fork+0x10/0x18
Code: d503201f d50332bf f94002a8 8b344108 (b9000113)
I can reliably reproduce this problem by running 'tail' on the registers
file in debugfs for the hdmi regmap.
# tail /sys/kernel/debug/regmap/62d87000.lpass-lpass_hdmi/registers
[ 84.658733] Unable to handle kernel paging request at virtual address ffffffd0128e800c
This crash happens because we're trying to read registers from the
regmap beyond the length of the mapping created by ioremap().
The number of hdmi_rdma_channels determines the size of the regmap via
this code in sound/soc/qcom/lpass-cpu.c:
lpass_hdmi_regmap_config.max_register = LPAIF_HDMI_RDMAPER_REG(variant, variant->hdmi_rdma_channels);
According to debugfs the size of the regmap is 0x68010 but according to
the DTS file posted in [1] the size is only 0x68000 (see the first reg
property of the lpass_cpu node). Let's change the number of channels to
be 3 instead of 4 so the math works out to have a max register of
0x67010, nicely fitting inside of the region size of 0x68000.
Note: I tried to bump up the size of the register region to the next
page to include the 0x68010 register but then the tail command caused
SErrors with an async abort, implying that the register region doesn't
exist or it isn't clocked because the bus is telling us that the
register read failed. I reduce the number of channels and played audio
through the HDMI channel and it kept working so I think this is correct.
Fixes: 2ad63dc8df ("ASoC: qcom: sc7180: Add support for audio over DP")
Link: https://lore.kernel.org/r/1601448168-18396-2-git-send-email-srivasam@codeaurora.org [1]
Cc: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Cc: Srinivasa Rao <srivasam@codeaurora.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20210115203329.846824-1-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS driver is partially broken on DragonBoard DB410c on 5.10 and
its totally broken on other Supported Qualcomm SoCs.
This was due to DAI ids being over written by the SoC specific header files
in the dt-bindings.
Idea of having SoC specific headers is not doable when we are dealing with
a common driver. So this patchset attempts to fix this properly by creating
a common dt-bindings header for lpass which can be updated with new entries
if required. This patchset also add an simple of_xlate function to resolve
the dai names and different SoCs might not have 1:1 mapping for the
dai_driver array with dai ids.
Changes since v1:
- removed array indexes as suggested by Stephan G.
- rebased to sound/for-next branch
- collected Srinivasa tested-by tag for sc7180 platform.
Thanks,
srini
Srinivas Kandagatla (2):
ASoC: dt-bindings: lpass: Fix and common up lpass dai ids
ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY
include/dt-bindings/sound/apq8016-lpass.h | 7 +++----
include/dt-bindings/sound/qcom,lpass.h | 15 +++++++++++++++
include/dt-bindings/sound/sc7180-lpass.h | 6 ++----
sound/soc/qcom/lpass-cpu.c | 22 ++++++++++++++++++++++
sound/soc/qcom/lpass-platform.c | 12 ++++++++++++
sound/soc/qcom/lpass-sc7180.c | 9 +++------
sound/soc/qcom/lpass.h | 2 +-
7 files changed, 58 insertions(+), 15 deletions(-)
create mode 100644 include/dt-bindings/sound/qcom,lpass.h
--
2.21.0
hdmi-codec is an optional property. The 2 patches fix DAI link binding
error when the property doesn't exist in DTS.
Tzung-Bi Shih (2):
ASoC: mediatek: mt8183-mt6358: ignore TDM DAI link by default
ASoC: mediatek: mt8183-da7219: ignore TDM DAI link by default
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 5 ++++-
sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 ++++-
2 files changed, 8 insertions(+), 2 deletions(-)
--
2.30.0.284.gd98b1dd5eaa7-goog
This series adds unit tests for ASoC topology.
First fix problems found when developing and running test cases and
then add tests implementation.
Tests themselves are quite simple and just call
snd_soc_tplg_component_load() with various parameters and check the
result. Tests themselves are described in more detail in commits
adding them.
Goal is to expand the amount of test cases in following patches.
Prerequisity for this patchset are 2 patches which have already been
sent:
https://lore.kernel.org/alsa-devel/20210114163602.911205-1-amadeuszx.slawinski@linux.intel.com/T/#t
Description on how typical test case itself works:
In order to load topology we need to have 3 things:
card, codec component & platform component.
In typical test case we register card and platform component and bind
to dummy codec. There are of course execeptions, when we want to
test behaviour of topology API when component or card is missing.
Note that this is bit different from typical scenario (in SOF and skylake
drivers) where card is registered by machine driver and component by
platform driver, as we register both when setting up test.
If you check the test case most of them have similar architecture of:
1.
/* run test */
ret = snd_soc_register_card(&kunit_comp->card);
if (ret != 0 && ret != -EPROBE_DEFER)
KUNIT_FAIL(test, "Failed to register card");
2.
ret = snd_soc_component_initialize(&kunit_comp->comp, &test_component, test_dev);
KUNIT_EXPECT_EQ(test, 0, ret);
3.
ret = snd_soc_add_component(&kunit_comp->comp, NULL, 0);
KUNIT_EXPECT_EQ(test, 0, ret);
Ad. 1.
First we register card, which in most tests returns -EPROBE_DEFER
(from snd_soc_bind_card()), as platform component is not yet created.
I test for both 0 and -EPROBE_DEFER, as it makes it easier to reshuffle
this code around if needed and there is one test case which does it in
different order.
Ad. 2.
Then we initialize platform component with structure pointing at proper
probe function, which calls snd_soc_tplg_component_load() with test
parameters and checks expected result.
Ad. 3.
And then in follow up we call snd_soc_add_component() which creates
platform component for us and calls snd_soc_try_rebind_card() which
if everything is bound properly calls previously set probe function.
Amadeusz Sławiński (5):
ASoC: topology: Properly unregister DAI on removal
Revert "ASoC: soc-devres: add devm_snd_soc_register_dai()"
ASoC: topology: KUnit: Add KUnit tests passing various arguments to
snd_soc_tplg_component_load
ASoC: topology: KUnit: Add KUnit tests passing empty topology with
variants to snd_soc_tplg_component_load
ASoC: topology: KUnit: Add KUnit tests passing topology with PCM to
snd_soc_tplg_component_load
include/sound/soc.h | 4 -
sound/soc/Kconfig | 17 +
sound/soc/Makefile | 5 +
sound/soc/soc-devres.c | 37 --
sound/soc/soc-topology-test.c | 843 ++++++++++++++++++++++++++++++++++
sound/soc/soc-topology.c | 9 +-
6 files changed, 870 insertions(+), 45 deletions(-)
create mode 100644 sound/soc/soc-topology-test.c
--
2.25.1
Since the recent refactoring, it's been reported that some USB-audio
devices (typically webcams) are no longer detected properly by
PulseAudio. The debug session revealed that it's failing at probing
by PA to try the sample rate 44.1kHz while the device has discrete
sample rates other than 44.1kHz. But the puzzle was that arecord
works as is, and some other devices with the discrete rates work,
either.
After all, this turned out to be the lack of the dependencies in a few
hw constraint rules: snd_pcm_hw_rule_add() has the (variable)
arguments specifying the dependent parameters, and some functions
didn't set the target parameter itself as the dependencies. This
resulted in an invalid parameter that could be generated only in a
certain call pattern. This bug itself has been present in the code,
but it didn't trigger errors just because the rules were casually
avoiding such a corner case. After the recent refactoring and
cleanup, however, the hw constraints work "as expected", and the
problem surfaced now.
For fixing the problem above, this patch adds the missing dependent
parameters to each snd_pcm_hw_rule() call.
Fixes: bc4e94aa8e ("ALSA: usb-audio: Handle discrete rates properly in hw constraints")
BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014
Link: https://lore.kernel.org/r/20210120204554.30177-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdmi-codec is an optional property. Ignore to bind TDM DAI link
if the property isn't specified.
Fixes: 5bdbe97711 ("ASoC: mediatek: mt8183-da7219: use hdmi-codec")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210120092237.1553938-3-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
hdmi-codec is an optional property. Ignore to bind TDM DAI link
if the property isn't specified.
Fixes: f2024dc55f ("ASoC: mediatek: mt8183: use hdmi-codec")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210120092237.1553938-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAIs need to be removed when topology unload function is called (usually
done when component is being removed). We can't do this when device is
being removed, as structures we operate on when removing DAI can already
be freed.
Fixes: 6ae4902f2f ("ASoC: soc-topology: use devm_snd_soc_register_dai()")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210120152846.1703655-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The allocation uses sizeof(u32) when it should use sizeof(unsigned long)
so it leads to memory corruption later in the function when the data is
initialized.
Fixes: 5aebe7c7f9 ("ASoC: topology: fix endianness issues")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/YAf+8QZoOv+ct526@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
BIT_WIDTH field in I2S_CTL register is two bits wide, however
recent regmap field conversion patch trimmed it down to one bit.
Fix this by correcting the bit range!
Fixes: b5022a36d2 ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210119174700.32639-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
lpass hdmi support patch totally removed support for MI2S TERTIARY
and QUATERNARY.
One of the major issue was spotted with the design of having
separate SoC specific header files for the common lpass driver.
This design is prone to break as an when new SoC header is added
as the common DAI ids of other SoCs will be overwritten by the
new ones.
Having a common header qcom,lpass.h should fix the issue and any new
DAI ids should be added to the common header.
With this change lpass also needs a new of_xlate function to resolve
dai name.
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Reported-by: Jun Nie <jun.nie@linaro.org>
Reported-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivasa Rao <srivasam@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20210119171527.32145-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
After hibernation, HDA controller can't be runtime-suspended after
commit 215a22ed31 ("ALSA: hda: Refactor codjc PM to use
direct-complete optimization"), which enables direct-complete for HDA
codec.
The HDA codec driver didn't expect direct-complete will be disabled
after it returns a positive value from prepare() callback. However,
there are some places that PM core can disable direct-complete. For
instance, system hibernation or when codec has subordinates like LEDs.
So if the codec is prepared for direct-complete but PM core still calls
codec's suspend or freeze callback, partially revert the commit and take
the original approach, which uses pm_runtime_force_*() helpers to
ensure PM refcount are balanced. Meanwhile, still keep prepare() and
complete() callbacks to enable direct-complete and request a resume for
jack detection, respectively.
Reported-by: Kenneth R. Crudup <kenny@panix.com>
Fixes: 215a22ed31 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210119152145.346558-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes for v5.11, mostly around HDA jack detection, plus
a couple of updates to the MAINTAINERS entries.
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Merge tag 'asoc-fix-v5.11-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.11
A few more fixes for v5.11, mostly around HDA jack detection, plus
a couple of updates to the MAINTAINERS entries.
When switching between firmware types, the wrong control
can be selected when requesting control in kernel API.
Use the currently selected DSP firwmare type to select
the proper mixer control.
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210115201105.14075-1-james.schulman@cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For addressing the regression on Pioneer devices, we recently
corrected the quirk code to enable the implicit feedback mode on those
devices properly. However, the devices still showed problems with the
full duplex operations with JACK, and after debug sessions, we figured
out that the older kernels that had worked with JACK also didn't use
the implicit feedback mode at all although they had the quirk code to
enable it; instead, the old code worked just to skip the normal sync
endpoint setup that would have been detected without it. IOW, what
broke without the implicit-fb quirk in the past was the application of
the normal sync endpoint that is actually the capture data endpoint on
these devices.
This patch covers the overseen piece: it modifies the quirk code again
not to enable the implicit feedback mode but just to make the driver
skipping the sync endpoint detection. This made the driver working
with JACK full-duplex mode again.
Still it's not quite clear why the implicit feedback doesn't work on
those devices yet; maybe it's about some issues in the URB setup. But
at least, with this patch, the driver should work in the level of the
older kernels again.
Fixes: 167c9dc84e ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices")
Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2/3 sample rate setup is based on the clock node, which is
usually shared in the interface, and can't be re-setup without
deselecting the interface once, and that's how the current code
behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence
we basically need to call for each endpoint usage even if those share
the same interface.
This patch fixes the behavior of UAC1 to call always
snd_usb_init_sample_rate() in snd_usb_endpoint_configure().
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint. Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.
This patch simplifies the code and corrects the target endpoint as
described in the above. It simply refers to fmt->endpoint directly.
Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.
Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add HD Audio Device PCI ID for the Intel Cometlake-R platform
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh@canonical.com>
Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_oss_synth_make_info() didn't check the error code from
snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy()
with the uninitialized string as the source, which may lead to the
access over the limit.
Add the proper error check for avoiding the failure.
Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Apire E5-575T laptop with codec ALC255 has a terrible
background noise comes from internal mic capture. And the jack
sensing dose not work for headset like some other Acer laptops.
This patch limits the internal mic boost on top of the existing
ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup. This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams. This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.
This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync. The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead. This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.
Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.
Fixes: 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The earlier commit to fix runtime PM in case i915 init fails,
introduces a possibility to hit a page fault.
snd_hdac_ext_bus_device_exit() is designed to be called from
dev.release(). Calling it outside device reference counting, is
not safe and may lead to calling the device_exit() function
twice. Additionally, as part of ext_bus_device_init(), the device
is also registered with snd_hdac_device_register(). Thus before
calling device_exit(), the device must be removed from device
hierarchy first.
Fix the issue by rolling back init actions by calling
hdac_device_unregister() and then releasing device with put_device().
This matches with existing code in hdac-ext module.
To complete the fix, add handling for the case where
hda_codec_load_module() returns -ENODEV, and clean up the hdac_ext
resources also in this case.
In future work, hdac-ext interface should be extended to allow clients
more flexibility to handle the life-cycle of individual devices, beyond
just the current snd_hdac_ext_bus_device_remove(), which removes all
devices.
BugLink: https://github.com/thesofproject/linux/issues/2646
Reported-by: Jaroslav Kysela <perex@perex.cz>
Fixes: 6c63c954e1 ("ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Link: https://lore.kernel.org/r/20210113150715.3992635-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
System takes a very long time to suspend after commit 215a22ed31
("ALSA: hda: Refactor codec PM to use direct-complete optimization"):
[ 90.065964] PM: suspend entry (s2idle)
[ 90.067337] Filesystems sync: 0.001 seconds
[ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done.
[ 90.188713] OOM killer disabled.
[ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done.
[ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug)
[ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend
[ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 329.490933] ACPI: EC: interrupt blocked
That commit keeps the codec suspended during the system suspend. However,
mute/micmute LED will clear codec's direct-complete flag by
dpm_clear_superiors_direct_complete().
This doesn't play well with SOF driver. When its runtime resume is
called for system suspend, hda_codec_jack_check() schedules
jackpoll_work which uses snd_hdac_is_power_on() to check whether codec
is suspended. Because the direct-complete path isn't taken,
pm_runtime_disable() isn't called so snd_hdac_is_power_on() returns
false and jackpoll continues to run, and snd_hda_power_up_pm() cannot
power up an already suspended codec in multiple attempts, causes the
long delay on system suspend:
if (dev->power.direct_complete) {
if (pm_runtime_status_suspended(dev)) {
pm_runtime_disable(dev);
if (pm_runtime_status_suspended(dev)) {
pm_dev_dbg(dev, state, "direct-complete ");
goto Complete;
}
pm_runtime_enable(dev);
}
dev->power.direct_complete = false;
}
When direct-complete path is taken, snd_hdac_is_power_on() returns true
and hda_jackpoll_work() is skipped by accident. So this is still not
correct.
If we were to use snd_hdac_is_power_on() in system PM path,
pm_runtime_status_suspended() should be used instead of
pm_runtime_suspended(), otherwise pm_runtime_{enable,disable}() may
change the outcome of snd_hdac_is_power_on().
Because devices suspend in reverse order (i.e. child first), it doesn't
make much sense to resume an already suspended codec from audio
controller. So avoid the issue by making sure jackpoll isn't used in
system PM process.
Fixes: 215a22ed31 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-3-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify hda_codec_jack_wake_enable() to also support disable WAKEEN.
In addition, this patch also moves the WAKEEN disablement call out of
hda_codec_jack_check() into hda_codec_jack_wake_enable().
This is a preparation for next patch.
No functional change intended.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-2-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of queueing jackpoll_work, runtime resume the codec to let it
use different jack detection methods based on jackpoll_interval.
This partially matches SOF driver's behavior with commit a6e7d0a4bd
("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the
difference is SOF unconditionally resumes the codec.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.
Reviewed-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As snd_fw_async_midi_port.consume_bytes is unsigned int, and
NSEC_PER_SEC is 1000000000L, the second multiplication in
port->consume_bytes * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes port->consume_bytes <= 16777.
Fixes: 531f471834 ("ALSA: firewire-lib/firewire-tascam: localize async midi port")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L,
the second multiplication in
ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes ff->rx_bytes[port] <= 16777.
Fixes: 1917429578 ("ALSA: fireface: add transaction support")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently hda on tegra30 fails to open a stream with an input/output error.
For example:
speaker-test -Dhw:0,3 -c 2
speaker-test 1.2.2
Playback device is hw:0,3
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 64 to 16384
Period size range from 32 to 8192
Using max buffer size 16384
Periods = 4
was set period_size = 4096
was set buffer_size = 16384
0 - Front Left
Write error: -5,Input/output error
xrun_recovery failed: -5,Input/output error
Transfer failed: Input/output error
The tegra-hda device was introduced in tegra30 but only utilized in
tegra124 until recent chips. Tegra210/186 work only due to a hardware
change. For this reason it is unknown when this issue first manifested.
Discussions with the hardware team show this applies to all current tegra
chips. It has been resolved in the tegra234, which does not have hda
support at this time.
The explanation from the hardware team is this:
Below is the striping formula referenced from HD audio spec.
{ ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
The current issue is seen because Tegra HW has a problem with boundary
condition (= 8) for striping. The reason why it is not seen on
Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is
read from GCAP register.
For the given stream (channels = 2, bps = 16);
ratio = (channels * bps) / NSDO = 32 / NSDO;
On Tegra30, ratio = 32/4 = 8 (FAIL)
On Tegra210/186, ratio = 32/2 = 16 (PASS)
On Tegra194, ratio = 32/4 = 8 (FAIL) ==> Earlier workaround was
applied for it
If Tegra210/186 is forced to use 4SDO, it fails there as well. So the
behavior is consistent across all these chips.
Applying the fix in [1] universally resolves this issue on tegra30-hda.
Tested on the Ouya game console and the tf201 tablet.
[1] commit 60019d8c65 ("ALSA: hda/tegra: workaround playback failure on
Tegra194")
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Ion Agorria <ion@agorria.com>
Reviewed-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Peter Geis <pgwipeout@gmail.com>
Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback. Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.
This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
