As the runtime PM issue was addressed by the recent fix 4961167bf7
("ALSA: hda/via: Apply the workaround generically for Clevo machines")
for VIA codecs, we need no longer to keep the Clevo device off from
the power saving as default. Drop the deny list entry accordingly.
Depends: 4961167bf7 ("ALSA: hda/via: Apply the workaround generically for Clevo machines")
Link: https://lore.kernel.org/r/20210202092744.20321-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with most Pioneer devices, the device descriptor is vendor specific
and as such, the number of channels, the PCM format, endpoints and
sample rate need to be specified. This device has 8 inputs and 8 outputs
and a sample rate of 48000 only. The PCM format is S24_3LE like other
devices.
There seems to be an appetite for reducing duplication amongs these
Pioneer patches but again, I feel this is a step to be taken after
support has been added as it's not completely clear where the
commonalities are.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-3-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the DJM-750, ensure that the format control message is passed to
the device when opening a stream. It seems as though fmt->sync_ep is not
always set when this function is called hence the passing of the value
at the call site. If this can be fixed, fmt->sync_up should be used as
the wvalue.
There doesn't seem to be a "cpu_to_le24" type function defined hence for
the open code but I did see a similar thing done in Bluez lib. Perhaps
we can get these definitions defined in byteorder.h. See hci_cpu_to_le24
in include/net/bluetooth/hci.h:2543 for similar usage.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-2-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Had a typo in lpass platform driver that resulted in crash
during suspend/resume with an HDMI dongle connected.
The regmap read/write/volatile regesters validation callbacks in lpass-cpu
were using MI2S rdma_channels count instead of hdmi_rdma_channels.
This typo error causing to read registers from the regmap beyond the length
of the mapping created by ioremap().
This fix avoids the need for reducing number hdmi_rdma_channels,
which is done in
commit 7dfe20ee92 ("ASoC: qcom: Fix number of HDMI RDMA channels on sc7180").
So reverting the same.
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/20210202062727.22469-1-srivasam@codeaurora.org
Reviewed-by: Stephen Boyd <swboyd@chromium.org>
Tested-by: Stephen Boyd <swboyd@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
When creating topology templates and overriding data in specific test
cases it should be done with cpu_to_le32 macro, so we operate on correct
data on all architectures, as topology parser use le32_to_cpu to parse
data from structures.
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20210202163123.3942040-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DT properties "dmic-mode" and "mic-type-X" are optional. Reduces the
log verbosity and changes the message a bit to avoid misleading.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210202033557.1621029-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixup BE DAI links channel count to match topology settings. Normally the
channel count of BE is equal to FE's so we don't have any issue. For some
cases like DSM with 2-channel FE and 4-channel BE the mismatch of BE and
topology will result in audio issues.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Keyon Jie <yang.jie@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210201092345.1214232-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All amba drivers return 0 in their remove callback. Together with the
driver core ignoring the return value anyhow, it doesn't make sense to
return a value here.
Change the remove prototype to return void, which makes it explicit that
returning an error value doesn't work as expected. This simplifies changing
the core remove callback to return void, too.
Reviewed-by: Ulf Hansson <ulf.hansson@linaro.org>
Reviewed-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Acked-by: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org> # for drivers/memory
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Acked-by: Suzuki K Poulose <suzuki.poulose@arm.com> # for hwtracing/coresight
Acked-By: Vinod Koul <vkoul@kernel.org> # for dmaengine
Acked-by: Guenter Roeck <linux@roeck-us.net> # for watchdog
Acked-by: Wolfram Sang <wsa@kernel.org> # for I2C
Acked-by: Takashi Iwai <tiwai@suse.de> # for sound
Acked-by: Vladimir Zapolskiy <vz@mleia.com> # for memory/pl172
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20210126165835.687514-5-u.kleine-koenig@pengutronix.de
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
This change adds audio jack injection feature through debugfs, with
this feature, we could validate alsa userspace changes by injecting
plugin or plugout events to the non-phantom audio jacks.
With this change, the sound core will build the folders
$debugfs_mount_dir/sound/cardN if SND_DEBUG and DEBUG_FS are enabled.
And if users also enable the SND_JACK_INJECTION_DEBUG, the jack
injection nodes will be built in the folder cardN like below:
$tree $debugfs_mount_dir/sound
$debugfs_mount_dir/sound
├── card0
│ ├── HDMI_DP_pcm_10_Jack
│ │ ├── jackin_inject
│ │ ├── kctl_id
│ │ ├── mask_bits
│ │ ├── status
│ │ ├── sw_inject_enable
│ │ └── type
...
│ └── HDMI_DP_pcm_9_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── card1
├── HDMI_DP_pcm_5_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
...
├── Headphone_Jack
│ ├── jackin_inject
│ ├── kctl_id
│ ├── mask_bits
│ ├── status
│ ├── sw_inject_enable
│ └── type
└── Headset_Mic_Jack
├── jackin_inject
├── kctl_id
├── mask_bits
├── status
├── sw_inject_enable
└── type
The nodes kctl_id, mask_bits, status and type are read-only, users
could check jack or jack_kctl's information through them.
The nodes sw_inject_enable and jackin_inject are directly used for
injection. The sw_inject_enable is read-write, users could check if
software injection is enabled or not on this jack, and users could
echo 1 or 0 to enable or disable software injection on this jack. Once
the injection is enabled, the jack will not change by hardware events
anymore, once the injection is disabled, the jack will restore the
last reported hardware events to the jack. The jackin_inject is
write-only, if the injection is enabled, users could echo 1 or 0 to
this node to inject plugin or plugout events to this jack.
For the detailed usage information on these nodes, please refer to
Documentation/sound/designs/jack-injection.rst.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210127085639.74954-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Output loopback is a feature where you can record what you hear.
The HDSP series of the RME interfaces provides this functionality
at the hardware level and this patch exposes controls to enable or
disable it per output (playback) channel.
This probably works on other cards but due to a lack of hardware
it is only tested and enabled for the HDSP9632 card with this patch.
Should this patch be accepted a separate patch will be posted to
https://github.com/alsa-project/alsa-tools/tree/master/hdspmixer
which adds "LPBK" buttons to each output in the playback strip for
the user to be able to control this feature from the user land.
Users from Windows tool TotalMixFX should be familiar with this.
Signed-off-by: Jasmin Fazlic <superfassl@gmail.com>
Link: https://lore.kernel.org/r/95cb3117-e85a-51a6-c2ce-bf736e70fc4c@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct mask is 0x1f8 (Bit 3-8), but due to missing BIT() 0xf (Bit
0-3) was set instead. This means setting of CPCAP_BIT_MIC1_RX_TIMESLOT0
(Bit 3) still worked (part of both masks). On the other hand the code
does not properly clear the other MIC timeslot bits. I think this
is not a problem, since they are probably initialized to 0 and not
touched by the driver anywhere else. But the mask also contains some
wrong bits, that will be cleared. Bit 0 (CPCAP_BIT_SMB_CDC) should be
safe, since the driver enforces it to be 0 anyways.
Bit 1-2 are CPCAP_BIT_FS_INV and CPCAP_BIT_CLK_INV. This means enabling
audio recording forces the codec into SND_SOC_DAIFMT_NB_NF mode, which
is obviously bad.
The bug probably remained undetected, because there are not many use
cases for routing microphone to the CPU on platforms using cpcap and
user base is small. I do remember having some issues with bad sound
quality when testing voice recording back when I wrote the driver.
It probably was this bug.
Fixes: f6cdf2d344 ("ASoC: cpcap: new codec")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Reviewed-by: Tony Lindgren <tony@atomide.com>
Link: https://lore.kernel.org/r/20210123172945.3958622-1-sre@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio Graph Card based Tegra driver is only useful on NVIDIA Tegra SoCs.
Hence add a dependency on SND_SOC_TEGRA, to prevent asking the user
about this driver when configuring a kernel without Tegra sound support.
Wrap all Tegra sound config options inside a big if/endif block, instead
of just adding the dependency to the single config option that does not
have it yet, to preventing similar future mistakes.
Fixes: 202e2f7745 ("ASoC: tegra: Add audio graph based card driver")
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20210129125915.2652952-1-geert+renesas@glider.be
Signed-off-by: Mark Brown <broonie@kernel.org>
When there is no TLV data in topology, extracting the TLV data
could result in a NULL pointer exception. Prevent this by making
sure that the TLV data exists before extracting it.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210201093128.1226603-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When device_type == DEVICE_ALI, we should also check the return
value of pci_iomap() to avoid potential null pointer dereference.
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20210131100916.7915-1-dinghao.liu@zju.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only usage of hdac_dev_attr_group is to put its address in an array
of pointers to const attribute_group structs. Make it const to allow the
compiler to put it in read-only memory.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20210131001241.2278-3-rikard.falkeborn@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only usage of ac97_adapter_attr_group is to put its address in an
array of pointers to const attribute_group structs. Make it const to
allow the compiler to put it in read-only memory.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20210131001241.2278-2-rikard.falkeborn@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
microblaze-linux-gcc (GCC) 9.3.0 complains about missing __ffssi2
symbol while using __builtin_ffs at runtime.
This is because arch/h8300 is compiled with -fno-builtin option.
so fallback and use kernel ffs() instead to all the arch builds happy!
Fixes: 1da0b9899a ("ASoC: soc-component: add snd_soc_component_read/write_field()")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210129100539.23459-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Series to refactor the DSP core management:
- move tracking of powered up DSP cores to common SOF code
- add logic filter unnecessary power actions
- modify existing implementations to use common code
whenever DSP cores are powered, so the state in common
code stays in sync
Bard Liao (5):
ASoC: SOF: Intel: hda: use snd_sof_dsp_core_power_up/down API
ASoC: SOF: Intel: hda-loader: keep init cores alive
ASoC: SOF: update dsp core power status in common APIs
ASoC: SOF: Filter out unneeded core power up/downs
ASoC: SOF: intel: hda-loader: use snd_sof_dsp_core_power_down/up APIs
sound/soc/sof/intel/hda-dsp.c | 2 +-
sound/soc/sof/intel/hda-loader.c | 9 +++++----
sound/soc/sof/intel/hda.c | 2 +-
sound/soc/sof/loader.c | 6 ------
sound/soc/sof/ops.h | 24 ++++++++++++++++++------
sound/soc/sof/pm.c | 1 -
sound/soc/sof/topology.c | 8 --------
7 files changed, 25 insertions(+), 27 deletions(-)
base-commit: e32df14235
--
2.29.2
The snd-soc-acpi-intel-match has duplicate module tags for all
platforms separately. Remove all but one and save some storage
space and cleanup modinfo output.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210128105751.1049837-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To manage enabled_cores_mask flag, we should always use snd_sof_dsp_
core_power_down/up APIs to power up/down dsp cores. The APIs do
a little bit more than the original functions, but it is harmless.
Suggested-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210128093850.1041387-6-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Exclude cores that are already powered on/off correctly. This allows to
simplify dsp_power_up/down() implementations and avoid unexpected error.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210128093850.1041387-5-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Only manage enabled_cores_mask in common snd_sof_dsp_core_power_up/down
APIs to ensure it stays in sync with actual DSP core state.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210128093850.1041387-4-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
init_core_mask should be the available cores mask after fw boot. So we
should keep not core 0 but init cores alive.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210128093850.1041387-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To implement common logic in SOF core, core power up/down flows should
use common SOF API and not directly use low-level platform specific
helper functions.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210128093850.1041387-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cancel the D0i3 work during runtime suspend as no streams are
active at this point anyway.
Fixes: 63e51fd33f ("ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.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/20210128092345.1033085-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No need of BCLK state maintenance from driver side as
clock_enable and clk_disable API's maintaing state counter.
One of the major issue was spotted when Headset jack inserted
while playback continues, due to same PCM device node opens twice
for playaback/capture and closes once for capture and playback continues.
It can resolve the errors in such scenarios.
Fixes: b182496822 ("ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state")
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210127151824.8929-1-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The double negative makes it hard to read "if (!ACPI_FAILURE(status))".
Replace it with "if (ACPI_SUCCESS(status))".
Signed-off-by: Bjorn Helgaas <bhelgaas@google.com>
Acked-by: Guenter Roeck <linux@roeck-us.net>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
ADL will use sof-adl-s.ri if it is ADL-S platform. So let's use
the default_fw_filename in pdata->desc for the ADL FW filename.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210125070500.807474-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_put_volsw in max98373_feedback_get is a typo, change it
to snd_soc_get_volsw.
Fixes: 349dd23931 ("ASoC: max98373: don't access volatile registers in bias level off")
Signed-off-by: Judy Hsiao <judyhsiao@google.com>
Link: https://lore.kernel.org/r/20210127135620.1143942-1-judyhsiao@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
It's often the case that we would write or read a particular field
in register. With the current soc_component apis, reading a particular
field in register would involve first read the register and then
perform shift operations.
Ex:
to read from a field mask of 0xf0
val = snd_soc_component_read(component, reg);
field = ((val & 0xf0) >> 0x4);
This is sometimes prone to errors and code become less readable!
With this new api we could just do
field = snd_soc_component_read_field(component, reg, 0xf0);
this makes it bit simple, easy to write and less error prone!
This also applies to writing!
There are various places in kernel which provides such field interfaces
however soc_component seems to be missing this.
This patch is inspired by FIELD_GET/FIELD_PREP macros in include/linux/bitfield.h
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210126171749.1863-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Enabling DMI L1 for capture streams could result in xruns during
pause/release. As pause/release is not a valid scenario for trace,
we can safely enable DMI L1 for it.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20210127020737.1088960-3-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DMI L1 entry is currently disabled whenever any capture stream is
opened to prevent xruns during pause/release. But, in
order to maximise power savings for the wake-on-voice usecase,
DMI L1 entry should be enabled for D0i3-compatible capture streams.
Introduce a new field, flags in struct sof_intel_hda_stream
that stores whether a stream is dmi_l1_compatible. All playback streams,
and D0i3-compatible capture streams are DMI L1 compatible.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20210127020737.1088960-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SOF firmware and topology files are not distributed via
linux-firmware. To help debugging cases where correct firmware is
not installed, print a pointer to the official upstream repository
for Sound Open Firmware releases.
BugLink: https://github.com/thesofproject/sof/issues/3665
Reported-by: Bruce Perens <bruce@perens.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Marc Herbert <marc.herbert@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://lore.kernel.org/r/20210127122358.1014458-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
tgl_3_in_1_default link topology may be used by both TGL-LP and TGL-H.
Let's remove the sof_fw_filename setting in struct snd_soc_acpi_mach
and use the default_fw_filename setting in struct sof_dev_desc.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Tested-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210125070500.807474-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The old code always uses sof_fw_filename in struct snd_soc_acpi_mach
as the firmware name. However, firmware name should depend on the platform
instead of the machine. For example, different machines may use the same
soundwire link topology, but they are using the different firmware. In this
case, it's hard to determine in struct snd_soc_acpi_mach which firmware it
should use.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Tested-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210125070500.807474-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A const prefix was put wrongly in the middle at the code refactoring
commit 932eaf7c79 ("ASoC: sh: siu_pcm: remove snd_pcm_ops"), which
leads to a build error as:
sound/soc/sh/siu_pcm.c:546:8: error: expected '{' before 'const'
Also, another inconsistency is that the declaration of siu_component
misses the const prefix.
This patch corrects both failures.
Fixes: 932eaf7c79 ("ASoC: sh: siu_pcm: remove snd_pcm_ops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210126154702.3974-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
The driver core ignores the return value of the remove callback, so
don't give isa drivers the chance to provide a value.
Adapt all isa_drivers with a remove callbacks accordingly; they all
return 0 unconditionally anyhow.
Acked-by: Marc Kleine-Budde <mkl@pengutronix.de> # for drivers/net/can/sja1000/tscan1.c
Acked-by: William Breathitt Gray <vilhelm.gray@gmail.com>
Acked-by: Wolfram Sang <wsa@kernel.org> # for drivers/i2c/
Reviewed-by: Takashi Iway <tiwai@suse.de> # for sound/
Reviewed-by: Hans Verkuil <hverkuil-cisco@xs4all.nl> # for drivers/media/
Signed-off-by: Uwe Kleine-König <uwe@kleine-koenig.org>
Link: https://lore.kernel.org/r/20210122092449.426097-4-uwe@kleine-koenig.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "chip" can't be NULL in hda_tegra_runtime_resume() because code would
crash otherwise. Let's remove the unnecessary check in order to clean up
code a tad.
Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Suggested-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-4-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reset hardware on RPM-resume in order to bring it into a predictable
state.
Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-3-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use clk_bulk helpers to make code cleaner. Note that this patch changed
the order in which clocks are enabled to make code look nicer, but this
doesn't matter in terms of hardware.
Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-2-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers in ALSA firewire stack supports eventing to userspace
applications via ALSA hwdep interface. All of the drivers supports stream
lock events. Some of them supports their unique events according to
specification of target device.
ALSA bebob driver supports the stream lock event only. In the case, it's
enough to check condition only in loop with process blocking. However,
current implementation check it again after breaking the loop.
This commit removes the redundant check.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Reported-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210125140208.26318-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assert hardware resets before clocks are enabled and then de-assert them
after clocks are enabled. This brings hardware into a predictable state.
Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Dmitry Osipenko <digetx@gmail.com> # Nexus7 T30 audio works
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-7-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
AHUB driver misses D_AUDIO and APBIF resets. CPU hangs on trying to
access hardware if resets aren't de-asserted. This problem is currently
masked by the tegra-clk driver which implicitly de-asserts the resets when
the corresponding clocks are enabled. Soon the implicit de-assertion will
be gone from the tegra-clk driver, thus we need to fix the AHUB driver.
Add the missing resets to the driver.
Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Dmitry Osipenko <digetx@gmail.com> # Nexus7 T30 audio works
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-5-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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
Smatch complains that "count" is not clamped when "ff->dev_lock_changed"
and it leads to an information leak. Fortunately, that's not actually
possible and the condition can be deleted.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/YA6n6I8EcNAO5ZFs@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch complains that "count" isn't clamped properly and
"oxfw->dev_lock_changed" is false then it leads to an information
leak. But it turns out that "oxfw->dev_lock_changed" is always
set and the condition can be removed.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/YA6ntkBxT/4DJ4YK@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This adds support for the Nintendo 64 console's sound.
Signed-off-by: Lauri Kasanen <cand@gmx.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de>
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>
Topology API exposes just 2 function calls, to load and unload topology.
This adds sanity checks to make sure that it behaves well when some of
parameters are set incoorectly or not needed.
This makes developer live easier by failing early instead of proceeding
on and then failing in unexpected ways.
As loading and unloading topology usually happens one the overhead of
additional checks should be negligible.
Amadeusz Sławiński (2):
ASoC: topology: Ensure that needed parameters are set
ASoC: topology: Check if ops is set before dereference
sound/soc/soc-topology.c | 22 +++++++++++++++-------
1 file changed, 15 insertions(+), 7 deletions(-)
--
2.25.1
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
The application circuit shall provide MICVDD power.
In default, the codec driver doesn't need to enable LDO2.
In case, a board wants to use VBAT for micbias,
it should add a DAPM route which IN1P connects with LDO2 in the machine driver.
e.g. { "IN1P", NULL, "LDO2" },
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20210121100353.6402-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_params_symmetry() checks rate/channel/sample_bits state.
These are very similar but different, thus, it needs to have very
verbose code.
This patch use macro for it and make code more simple.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878s8un6si.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_apply_symmetry() want to call snd_pcm_hw_constraint_single()
for rate/channel/sample_bits, but, it needs many condition checks.
These are very similar but different, thus, it needs to have very
verbose code.
This patch use macro for it and make code more simple.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a6tan6sm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_runtime / snd_soc_dai / snd_soc_dai_driver / snd_soc_dai_link
have related parameter which is similar but not same naming.
struct snd_pcm_runtime {
...
(A) unsigned int rate;
...
(B) unsigned int sample_bits;
...
};
struct snd_soc_dai {
...
(A) unsigned int rate;
(B) unsigned int sample_bits;
...
};
struct snd_soc_dai_driver {
...
(A) unsigned int symmetric_rates:1;
(B) unsigned int symmetric_samplebits:1;
...
};
struct snd_soc_dai_link {
...
(A) unsigned int symmetric_rates:1;
(B) unsigned int symmetric_samplebits:1;
...
};
Because it is similar but not same naming rule,
code can be verbose / can't share macro.
This patch sync naming rule for framework.
- xxx_rates;
+ xxx_rate;
- xxx_samplebits;
+ xxx_sample_bits;
old name will be removed if all drivers were switched
to new naming rule.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wnweolj6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit a39748d03c ("ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry()")
was applied by miscommunication.
To more cleanup code, and to be easy review, this patch reverts it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2guoljm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology can be created without ops overrides, in that case trying to
assign any value would lead to dereferencing NULL pointer.
Other places in code have either checks for tplg->ops or loop using
*_count variables, hence they can't dereference NULL pointer and there
is no need to add more checks.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20210114163602.911205-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As snd_soc_tplg_component_load is exported function, which means it is
part of API, there should be checks if it is called with proper
parameters.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20210114163602.911205-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to ensure correct behaviour of topology API, add unit tests
exercising topology functionality.
Add topology containing PCM template and tests for parsing it. Also
adds test cases simulating modules reloads in case of separate drivers
for card and component.
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/20210120152846.1703655-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to ensure correct behaviour of topology API, add unit tests
exercising topology functionality.
Add "empty" topology template and tests for parsing it. Also adds few
variants with bad magic numbers.
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/20210120152846.1703655-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to ensure correct behaviour of topology API, add unit tests
exercising topology functionality.
Start with adding cases for passing various arguments to
snd_soc_tplg_component_load as it is part of exposed topology API.
First test case adds test passing NULL component as argument.
Following one adds test case for passing NULL ops as argument.
Finally add test case passing NULL fw as argument.
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/20210120152846.1703655-4-amadeuszx.slawinski@linux.intel.com
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>
At probing a UAC2/UAC3 device like NUX MG-300 USB interface, we get
error messages "RANGE setting not yet supported". It comes the place
where the driver tries to determine the resolution of mixer volumes
via SET_CUR_RES and GET_CUR_RES verbs. Those verbs aren't supported
on UAC2 and UAC3, hence the driver warns like the above. Although the
driver handles this error and works as expected, it's still ugly to
show such errors unnecessarily.
This patch papers over the errors by applying the resolution detection
only for UAC1 and skipping it for UAC2/UAC3.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current USB-audio driver gets an error at probing NUX MG-300 about
parsing the clocks. This is because the firmware doesn't return the
proper connection of the clock selector that is connected to a single
clock; it's likely that the firmware was lazy^w optimized and the
inquiry wasn't handled. Actually it makes little sense to inquire and
set up the single connection explicitly.
This patch fixes the issue by simply skipping the clock selector
inquiry if it's a single connection.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 1st and 2nd patches refactor the machine driver.
The 3rd patch changes the platform driver to support TDM 8 channel output.
The 4th patch adds an optional DT property.
The 5th patch makes the machine driver support DP audio if the optional DT
property is specified.
Tzung-Bi Shih (5):
ASoC: mediatek: mt8192-mt6359: move headset_jack to card specific data
ASoC: mediatek: mt8192-mt6359: simplify mt8192_rt5682_init
ASoC: mediatek: mt8192: change mclk_multiple of TDM from 128 to 512
ASoC: dt-bindings: mt8192-mt6359: add hdmi-codec property
ASoC: mediatek: mt8192-mt6359: support audio over DP
.../sound/mt8192-mt6359-rt1015-rt5682.yaml | 5 ++
sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 2 +-
.../mt8192/mt8192-mt6359-rt1015-rt5682.c | 54 ++++++++++++++++---
3 files changed, 52 insertions(+), 9 deletions(-)
--
2.30.0.284.gd98b1dd5eaa7-goog
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
The zte zx platform is getting removed, so this driver is no
longer needed.
Cc: Jun Nie <jun.nie@linaro.org>
Cc: Shawn Guo <shawnguo@kernel.org>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210120162553.21666-3-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The CSR SiRF prima2/atlas platforms are getting removed, so this driver
is no longer needed.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Barry Song <baohua@kernel.org>
Link: https://lore.kernel.org/r/20210120162553.21666-2-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The only way this driver can be probed is via devicetree, which always
provides driver data.
Remove the unneeded of_device_get_match_data() error check, as it
can never fail.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210118123815.1630882-6-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The only way this driver can be probed is via devicetree, which always
provides driver data.
Remove the unneeded of_device_get_match_data() error check, as it
can never fail.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210118123815.1630882-5-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The only way this driver can be probed is via devicetree, which always
provides driver data.
Remove the unneeded of_device_get_match_data() error check, as it
can never fail.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210118123815.1630882-4-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The retrieval of driver data via of_device_get_match_data() can make
the code simpler.
Use of_device_get_match_data() to simplify the code.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210118123815.1630882-2-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The retrieval of driver data via of_device_get_match_data() can make
the code simpler.
Use of_device_get_match_data() to simplify the code.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20210118123815.1630882-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If the DTS property is specified, the DP bridge should populate a
"hdmi-codec" platform device (sound/soc/codecs/hdmi-codec.c).
The "hdmi-codec" device is the communication relayer between the ASoC
machine driver and the DP bridge. For example:
- Notifies DP bridge when setting hw_param.
- Notifies ASoC when jack detection events.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210120080850.699354-6-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
mclk = rate * mclk_multiple
bclk = rate * channel * sample_width
If TDM outputs 8 channels and 32 bits, bclk will be greater than mclk.
Changes the ratio from 128 to 512.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210120080850.699354-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Returns snd_soc_component_set_jack() directly in mt8192_rt5682_init.
No need to have another block to check the return value.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20210120080850.699354-3-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: 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>
This series adds audio graph based sound card support for Tegra210
platforms like Jetson-TX1 an Jetson-Nano. The following preparatory
audio graph enhancement series is already merged.
* https://patchwork.kernel.org/project/alsa-devel/list/?series=375629&state=*
Following are the summary of changes:
* Add graph/audio-graph based schemas or schema updates for Tegra210
component and machine drivers.
* Add Tegra audio graph machine driver.
* Add required DT support for Jetson-TX1/Nano.
This work is based on earlier discussion of DPCM usage for Tegra
and simple card driver updates.
* https://lkml.org/lkml/2020/4/30/519
* https://lkml.org/lkml/2020/6/27/4
Original v6 series was sent about 6-7 weeks back. The dependency commit,
https://lore.kernel.org/alsa-devel/1610948585-16286-1-git-send-email-spujar@nvidia.com/
is now merged. Resending this now to appear in the top of the mail list.
Changelog
=========
v5 -> v6
--------
* Added ports or port description in YAML docs for Tegra AHUB
devices and graph card in patch 1/6 and 2/6. Reference of
audio-graph-port.yaml is used for AHUB devices.
* Dropped redundant NULL check return for of_device_get_match_data()
in patch 3/6.
* Added 'Reviewed-by' tag from Jon Hunter.
* No changes in remaining patches.
v4 -> v5
--------
* Audio graph related changes were sent in separate v5 series as
mentioned above and are dropped from current series.
* Graph and audio graph doc patches are dropped from this series
and are sent separately as mentioned above.
* Minor change with phandle label for TX1 and Nano platform DT files.
* No changes in other patches.
v3 -> v4
--------
* Added new patches to convert graph.txt and audio-graph-card.txt
to corresponding json-schema files. Later these references
are used in Tegra audio graph schema.
* AHUB component binding docs are updated to reflect the usage
of ports/port/endpoint
* More common stuff is moved into graph_parse_of() and this is
used by both generic and Tegra audio graph.
* DT binding for Tegra audio graph is updated to included "ports { }"
* As per the suggestion 'void *data' member is dropped from
'asoc_simple_priv' and instead container method is used to
maintain required custom data internal to Tegra audio graph.
v2 -> v3
--------
* Dropped new compatible addition in generic graph driver
after reviewing it with Morimoto-san. Instead added Tegra
audio graph driver and new compatibles are added in the same.
* Added new patches to expose new members for customization
in audio graph driver.
* Added new patch for Tegra audio graph driver and related
documentation.
* Minor change in below commit where mutex version of helper is used
"ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM"
* DT binding is updated to use the newly exposed compatibles
* No changes in other patches
v1 -> v2
--------
* Re-organized ports/endpoints description for ADMAIF and XBAR.
Updated DT patches accordingly.
* After above change, multiple Codec endpoint support is not
required and hence dropped for now. This will be considered
separately if at all required in future.
* Re-ordered patches in the series.
Sameer Pujar (6):
ASoC: dt-bindings: tegra: Add graph bindings
ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card
ASoC: tegra: Add audio graph based card driver
arm64: defconfig: Enable Tegra audio graph card driver
arm64: tegra: Audio graph header for Tegra210
arm64: tegra: Audio graph sound card for Jetson Nano and TX1
.../sound/nvidia,tegra-audio-graph-card.yaml | 187 +++++++++++++++
.../bindings/sound/nvidia,tegra186-dspk.yaml | 18 +-
.../bindings/sound/nvidia,tegra210-admaif.yaml | 13 +-
.../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +-
.../bindings/sound/nvidia,tegra210-dmic.yaml | 18 +-
.../bindings/sound/nvidia,tegra210-i2s.yaml | 18 +-
.../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++++++++
arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 +++++++++++++++++++++
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++++++++
arch/arm64/configs/defconfig | 1 +
sound/soc/tegra/Kconfig | 9 +
sound/soc/tegra/Makefile | 2 +
sound/soc/tegra/tegra_audio_graph_card.c | 251 ++++++++++++++++++++
13 files changed, 1085 insertions(+), 6 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml
create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi
create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c
--
2.7.4
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>
ASoC: cpcap: Implement set_tdm_slot for voice call support
For using cpcap for voice calls, we need to route audio directly from
the modem to cpcap for TDM (Time Division Multiplexing). The voice call
is direct data between the modem and cpcap with no CPU involvment. In
this mode, the cpcap related audio mixer controls work for the speaker
selection and volume though.
To do this, we need to implement standard snd_soc_dai_set_tdm_slot()
for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(),
snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a
voice call.
Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call()
helper to configure the additional registers needed for voice call.
Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for
CPCAP_BIT_VAUDIO_MODE0 set on init.
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Link: https://lore.kernel.org/r/20210112174704.GA13496@duo.ucw.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
It is not guaranteed that I2S RX is disabled when the kernel booting.
For example, if the kernel crashes while it is enabled, it will keep
enabled until the next time EC reboots. Reset I2S RX when probing to
fix this issue.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Reviewed-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20210115075301.47995-2-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Tegra audio machine driver which is based on generic audio graph card
driver. It re-uses most of the common stuff from audio graph driver and
uses the same DT binding. Required Tegra specific customizations are done
in the driver and additional DT bindings are required for clock handling.
Details on the customizations done:
- Update PLL rates at runtime: Tegra HW supports multiple sample rates
(multiples of 8x and 11.025x) and both of these groups require different
PLL rates. Hence there is a requirement to update this at runtime.
This is achieved by providing a custom 'snd_soc_ops' and in hw_param()
callback PLL rate is updated as per the sample rate.
- Internal structure 'tegra_audio_graph_data' is used to maintain clock
handles of PLL.
- The 'force_dpcm' flag is set to use DPCM for all DAI links.
- The 'component_chaining' flag is set to use DPCM with component model.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/1611048496-24650-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC tegra PCM code still has explicit calls of dma_set_mask() and
dma_set_coherent_mask().
Let's simplify with dma_set_mask_and_coherent().
Cc: Thierry Reding <thierry.reding@gmail.com>
Cc: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210114133337.1039-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC Intel SOF driver still has explicit calls of dma_set_mask() and
dma_set_coherent_mask().
Let's simplify with dma_set_mask_and_coherent().
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210114133337.1039-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC Intel Skylake driver still has explicit calls of dma_set_mask()
and dma_set_coherent_mask().
Let's simplify with dma_set_mask_and_coherent().
Cc: Cezary Rojewski <cezary.rojewski@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210114133337.1039-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
This adds the Pioneer DJ DJM-750 to the quirks table and ensures
skip_pioneer_sync_ep() is (also) called: this device uses the vendor
ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many
just like to be awkward it seems).
Playback on all 8 channels works. I'll likely keep this working in the
future and submit futher patches and improvements as necessary.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add PCI id for the AlderLake-P.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210114115558.52699-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The timeout for an individual transaction w/ the Cadence IP is the same as
the entire resume operation for codecs.
This doesn't make sense, we need to have at least one order of magnitude
between individual transactions and the entire resume operation.
Set the timeout on the Cadence side to 500ms and 5s for the codec resume.
Both ASoC and SoundWire trees are fine for this series.
Pierre-Louis Bossart (2):
ASoC: codecs: soundwire: increase resume timeout
soundwire: cadence: reduce timeout on transactions
drivers/soundwire/cadence_master.c | 2 +-
sound/soc/codecs/max98373-sdw.c | 4 +++-
sound/soc/codecs/rt1308-sdw.c | 2 +-
sound/soc/codecs/rt5682.h | 2 +-
sound/soc/codecs/rt700-sdw.c | 2 +-
sound/soc/codecs/rt711-sdw.c | 2 +-
sound/soc/codecs/rt715-sdw.c | 2 +-
7 files changed, 9 insertions(+), 7 deletions(-)
--
2.17.1
Here's some minor code cleanups for the lpass-cpu driver. I noticed that
it casts away const from the driver data from DT. That's not great but
fixing it is a little more involved. I'll get to it later. There's also
some questionable clk_get() usage that should probably be
clk_get_optional(). For now this should help a little.
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>
Stephen Boyd (4):
ASoC: qcom: Remove useless debug print
ASoC: qcom: Add some names to regmap configs
ASoC: qcom: Stop casting away __iomem for error pointers
ASoC: qcom: Remove duplicate error messages on ioremap
sound/soc/qcom/lpass-cpu.c | 17 ++++++-----------
1 file changed, 6 insertions(+), 11 deletions(-)
base-commit: 5c8fe583cc
--
https://chromeos.dev
The resume operation relies on multiple transactions to synchronize
the regmap state, make sure the timeout is one order of magnitude
larger than an individual transaction, so that timeouts of failed
transactions are detected first.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20210115061651.9740-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need to print an error message when these ioremap operations
fail. The function that returns an error already prints an error message
and properly attributes it to the device. Drop them to save some code.
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>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210115034327.617223-5-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need to cast away __iomem when testing with IS_ERR() or
converting with PTR_ERR(). Modern sparse can handle this just fine.
Drop it.
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>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210115034327.617223-4-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can sometimes have multiple regmaps. Let's add a name so
that we can differentiate in debugfs more easily.
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>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210115034327.617223-3-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This looks like a left over debug print that tells us that HDMI is
enabled. Let's remove it as that's definitely not an error to have HDMI
enabled.
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>
Fixes: 7cb37b7bd0 ("ASoC: qcom: Add support for lpass hdmi driver")
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210115034327.617223-2-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPE driver still has explicit calls of dma_set_mask() and
dma_set_coherent_mask().
Let's simplify with dma_set_mask_and_coherent().
Link: https://lore.kernel.org/r/20210114125412.993-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many PCI drivers still have two explicit calls of dma_set_mask() and
dma_set_coherent_mask().
Let's simplify with dma_set_mask_and_coherent().
Link: https://lore.kernel.org/r/20210114125412.993-2-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>
The last remaining usage of strlcpy() in USB-audio driver is the setup
of the card longname string. Basically we need to know whether any
non-empty string is set or not, and no real length is needed.
Refactor the code and use strscpy() instead. After this change,
strlcpy() is gone from all sound/* code.
Link: https://lore.kernel.org/r/20210115100437.20906-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver still contains two calls of strlcpy() because the
return size is evaluated. Basically it just checks whether the string
is copied or not, but since strcpy() may return a negative error code,
we should check the negative value and treat as filled.
Link: https://lore.kernel.org/r/20210115095758.19707-1-tiwai@suse.de
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>
Invoke hda_dsp_remove() as the .shutdown() callback. This will help to
perform shutdown of the DSP safely on TGL platforms before shutting down
or rebooting the system.
BugLink: https://github.com/thesofproject/linux/issues/2571
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-4-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the .shutdown() callback to the sof-pci-dev driver, to help to
handle shutting down specific tasks for SOF PCI platforms.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper snd_sof_device_shutdown() to wrap the platform specific
.shutdown callbacks for SOF platforms.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add .shutdown() callback to the struct snd_sof_dsp_ops, for
doing platform specific actions at shutdown.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210113152617.4048541-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Hi Mark
These are not so important, but for
soc-pcm cleanup patches.
Kuninori Morimoto (6):
ASoC: soc-pcm: move dpcm_set_fe_update_state()
ASoC: soc-pcm: add dpcm_set_be_update_state()
ASoC: soc-pcm: add soc_pcm_set_dai_params()
ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry()
ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()
ASoC: soc-pcm: setup pcm at one place in soc_new_pcm()
sound/soc/soc-pcm.c | 231 +++++++++++++++++---------------------------
1 file changed, 90 insertions(+), 141 deletions(-)
--
2.25.1
Thank you for your help !!
Best regards
---
Kuninori Morimoto
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>
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-3-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20210106133650.13509-2-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of manually managing its DMA buffers using
dma_{alloc,free}_coherent() lets the sound core take care of this using
managed buffers.
On one hand this reduces the amount of boiler plate code, but the main
motivation for the change is to use the shared code where possible. This
makes it easier to argue about correctness and that the code does not
contain subtle bugs like data leakage or similar.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Link: https://lore.kernel.org/r/20210106133650.13509-1-lars@metafoo.de
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment it is necessary to set up the DAPM routes between
front-end AIF<->DAI explicitly in the device tree, e.g. using
audio-routing =
"MM_DL1", "MultiMedia1 Playback",
"MM_DL3", "MultiMedia3 Playback",
"MM_DL4", "MultiMedia4 Playback",
"MultiMedia2 Capture", "MM_UL2";
This is prone to mistakes and (sadly) there is no clear error if one
of these routes is missing. :(
Actually, this should not be necessary because the ASoC core normally
automatically links AIF<->DAI within snd_soc_dapm_link_dai_widgets().
This is done using the "stname" parameter of SND_SOC_DAPM_AIF_IN/OUT.
For SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
it should create the route from above: MM_DL1 <-> MultiMedia1 Playback.
This does not work at the moment because the AIF widget (MM_DL1)
and the DAI widget (MultiMedia1 Playback) belong to different
DAPM contexts (q6routing / q6asm-dai).
Fix this by declaring the AIF widgets in the same driver as the DAIs
(q6asm-dai). Now the routes above are created automatically
and no longer need to be specified in the device tree.
This is also more consistent with the back-end AIFs which are already
declared in q6afe-dais instead of q6routing. q6routing should only link
the components together using mixers.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20201211203255.148246-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two issues with this code. The first error path forgot to set
the error code and instead returns success. The second error path
doesn't clean up.
Fixes: 272b5edd3b ("ASoC: Add support for CS42L56 CODEC")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X9NE/9nK9/TuxuL+@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_apply_symmetry() want to call snd_pcm_hw_constraint_single()
for rate/channel/sample_bits, but, it needs many condition check.
These are very similar but different, thus, it needs to have very
verbose code.
This patch use macro for it and make code more simple.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wnxo7uyq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Getting rate/channels/sample_bits from param needs fixed method.
This patch adds new soc_pcm_set_dai_params() and replace existing code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2i47uyw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-pcm has dpcm_set_fe_update_state() to update FE's runtime_update
(except dpcm_fe_dai_do_trigger() which needs to update it without it).
OTOH, it doesn't have BE's update function.
O: dpcm_set_fe_update_state()
X: dpcm_set_be_update_state()
This patch add BE's dpcm_set_fe_update_state()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zh2k7uz1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves dpcm_set_fe_update_state() to top side.
This is prepare for cleanup soc-pcm.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rfw99jn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi All,
This series adds support for devices with only a headphone jack
(no speakers/internal mic). Specifically this adds support for the
Mele PCG03 Mini PC. But the new no-speakers and no-internal-mic quirks
will likely be useful on other devices too.
Regards,
Hans
There could be more than one thread read/write the dsp_power_state
simultaneously (e.g. hda_dsp_d0i3_work and sof_ipc_tx_message), add a
mutex power_state_access to make sure the access to it is mutually
exclusive.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210105155640.3725238-1-kai.vehmanen@linux.intel.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>
sound/soc/soc-core.c: soc_remove_component() unconditionally calls
snd_soc_component_set_jack(component, NULL, NULL); on any components
being removed.
This means that on machines where the machine-driver does not provide
a jack through snd_soc_component_set_jack() es8316_disable_jack_detect()
will still get called and at this time es8316->jack will be NULL and
the es8316->jack->status check in es8316_disable_jack_detect() will
lead to a NULL pointer deref.
Fix this by checking for es8316->jack bein NULL at the start of
es8316_disable_jack_detect() and turn the function into a no-op in
that case.
Cc: russianneuromancer <russianneuromancer@ya.ru>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210112101725.44200-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20210108084456.6603-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the snd_soc_dai_set_tdm_slot() call from cht_codec_init() to
cht_codec_fixup(). There are 2 reasons for doing this:
1. This aligns the cht_bsw_nau8824 with all the other BYT/CHT machine
drivers which also do this from their codec_fixup function.
2. When using the SOF driver, things like the TDM info is set from the
topology file. Moving the call to the codec_fixup function, which gets
skipped when using the SOF driver avoids the call interfering with the
settings when using the SOF driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210107115324.11602-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the Mele PCG03 Mini PC, being a Mini PC this device
has no speakers and no internal microphone.
To make matters worse the speaker output pins are shorted (to gnd or
to each other?) and SPKVDD is provided. So trying to output sound on the
speakers leads to shorting SPKVDD, this leads to a power dip after
which the codec is an unknown state. Sometimes it drops of the i2c
bus, sometimes it does still respond to i2c transfers, but is otherwise
not functional. TL;DR: trying to use the speaker outputs on this model
is BAD.
Besides not having speakers / an internal mic, this is a Bay Trail CR
device without a CHAN package in ACPI, so we default to SSP0-AIF2 as
codec connection. But the device is actually using SSP0-AIF1, so we
need to quirk that too.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices, like mini PCs/media/top-set boxes do not have an internal
microphone at all, an example of the is the Mele PCG03 Mini PC.
Add a new BYT_RT5640_NO_INTERNAL_MIC_MAP input-mapping for this,
which does not add any internal-mic routes and modifies the components
and the (optional) long_name strings to reflect this.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices, like mini PCs/media/top-set boxes do not have any speakers
at all, an example of the is the Mele PCG03 Mini PC.
Add a new BYT_RT5640_NO_SPEAKERS quirk-flag which when sets does not add
speaker routes and modifies the components and the (optional) long_name
strings to reflect that there are no speakers.
Cc: Rasmus Porsager <rasmus@beat.dk>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210109210119.159032-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
These patches are trying to fix the jack detection and internal
microphone problems on ECS EF20 series laptops which are empowered
by Intel Atom x5-Z8350 CPU (CherryTrail) with Realtek rt5645 audio
codec.
---
v2 -> v3:
Restore the accidentally removed terminator of the
dmi_platform_data[].
v1 -> v2:
Invoke callback() of the DMI quirk if it exists, because
the dmi_first_match() doesn't.
---
Chris Chiu (4):
ASoC: rt5645: Introduce mapping for ACPI-defined GPIO
ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series
ASoC: rt5645: add inv_hp_det flag
ASoC: rt5645: Enable internal microphone and JD on ECS EF20
include/sound/rt5645.h | 2 ++
sound/soc/codecs/rt5645.c | 45 +++++++++++++++++++++++++++++++++++++++
2 files changed, 47 insertions(+)
--
2.20.1
The error path here doesn't set "ret" so it returns uninitialized data
instead of a negative error code.
Fixes: 2c1382840c ("ASoC: soc-pcm: disconnect BEs if the FE is not ready")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/X/wfXQFxeMLvpO+1@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
wm_adsp_read_data_word() used if (ret) to check for an error from
wm_adsp_read_raw_data_block(). While this is perfectly valid,
wm_adsp_read_raw_data_block() itself uses if (ret < 0) and three
calls to wm_adsp_read_data_word() also use if (ret < 0).
This creates an error check chain like this:
1st) if (ret < 0) return ret;
2nd) if (ret) return ret;
3rd) if (ret < 0) ...
This can confuse the compiler into thinking that there are possible
returns > 0 from the middle if() that are not handled by the final
if(). If this was true it would lead to using uninitialized variables
later in the outer function.
Fix this by changing the test in wm_adsp_read_data_word() to be
if (ret < 0).
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20210111133825.8758-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Kernel test robot throws below error ->
sound/soc/soc-pcm.c:2523 dpcm_run_update_startup() error: uninitialized
symbol 'ret'.
Initializing ret = 0 and returning correct -ERRNO in failure path.
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Souptick Joarder <jrdr.linux@gmail.com>
Link: https://lore.kernel.org/r/1610163901-5523-1-git-send-email-jrdr.linux@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will be the workaround to fix getting the wrong device ID on the rare chance.
It seems like something unstable when the system resumes. e.g. the bus clock
This patch tries to read the device ID to check several times.
After the test, the driver will get the correct device ID the second time.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20210111092740.9128-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On ECS EF20 series laptops, the internal mic is on DMIC2/IN2P.
And they need the inv_hp_det to make jack detection to work as
exoected.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-5-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The ECS EF20EA laptop use gpio for jack detection instead of rt5645
rt5645 JD. However, the GPIO polarity is inverse for hp-detect based
on the _DSD property of the RTK2 device.
Name (_DSD, Package () {
ToUUID("daffd814-6eba-4d8c-8a91-bc9bbf4aa301"),
Package () {
Package () {"hp-detect-gpio", Package() {^RTK2, 0, 0, 1 }},
}
})
This flag will invert the hp-detect gpio polarity.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-4-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
On at least one laptop (ECS EF20EA) the 'hp-detect' GPIO is defined in
the DSDT table by the ACPI GpioIo resources in _CRS. The GPIO related
information should be mapped to the rt5645 driver to enable the jack
detection also on non-DT platforms.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Link: https://lore.kernel.org/r/20210111054141.4668-2-chiu@endlessos.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset. The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset. This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode. For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.
This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint. The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.
Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The implicit feedback mode needs to handle two endpoints and the
choice of the audioformat object for the sync EP is important since
this determines the compatibility of the hw_params. The current code
uses the same audioformat object if both the main EP and the sync EP
point to the same iface/altsetting. This was done in consideration of
the non-implicit-fb sync EP handling, and it doesn't match well with
the cases where actually to endpoints are defined in the sameiface /
altsetting like a few Pioneer devices.
Modify snd_usb_find_implicit_fb_sync_format() to pick up the
audioformat that is assigned in the counter-part substreams primarily,
so that the actual capture stream can be opened properly. We keep the
same audioformat object only as a fallback in case nothing found,
though.
Fixes: 9fddc15e80 ("ALSA: usb-audio: Factor out the implicit feedback quirk code")
Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change in the endpoint management moved the endpoint object
creation from the stream open time to the parser of the audio
descriptor. It works fine for the standard audio, but it overlooked
the other places that create audio streams via quirks
(QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices;
those call snd_usb_add_audio_stream() manually, hence they miss the
endpoints, eventually resulting in the error at opening streams.
Moreover, now the sync EP setup was moved to the explicit call of
snd_usb_audioformat_set_sync_ep(), and this needs to be added for
those places, too.
This patch addresses those regressions for quirks. It adds a local
helper function add_audio_stream_from_fixed_fmt(), which does the all
needed tasks, and replaces the calls of snd_usb_add_audio_stream()
with this new function.
Fixes: 54cb31901b ("ALSA: usb-audio: Create endpoint objects at parsing phase")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable DMA transfer mode for Intel Keem Bay ASoC platform driver.
The driver will search the device tree for DMA resources at boot
time to enable DMA transfer mode, and will proceed to use DMA
transfer if the resource is available, otherwise the default PIO
mode will be used.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210108031248.20520-6-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.
Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20210108085834.7168-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Sound is broken on the DragonBoard 410c (apq8016_sbc) since 5.10:
hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95
qcom-apq8016-sbc 7702000.sound: Failed to set jack: -95
ADV7533: ASoC: error at snd_soc_link_init on ADV7533: -95
hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95
qcom-apq8016-sbc: probe of 7702000.sound failed with error -95
This happens because apq8016_sbc calls snd_soc_component_set_jack() on
all codec DAIs and attempts to ignore failures with return code -ENOTSUPP.
-ENOTSUPP is also excluded from error logging in soc_component_ret().
However, hdmi_codec_set_jack() returns -E*OP*NOTSUPP if jack detection
is not supported, which is not handled in apq8016_sbc and soc_component_ret().
Make it return -ENOTSUPP instead to fix sound and silence the errors.
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 55c5cc63ab ("ASoC: hdmi-codec: Use set_jack ops to set jack")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210107165131.2535-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
strlcpy is deprecated. see: Documentation/process/deprecated.rst
Change the calls that do not use the strlcpy return value to the
preferred strscpy.
Done with cocci script:
@@
expression e1, e2, e3;
@@
- strlcpy(
+ strscpy(
e1, e2, e3);
This cocci script leaves the instances where the return value is
used unchanged.
After this patch, sound/ has 3 uses of strlcpy() that need to be
manually inspected for conversion and changed one day.
$ git grep -w strlcpy sound/
sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname));
sound/usb/mixer.c: return strlcpy(buf, p->name, buflen);
sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen);
Miscellenea:
o Remove trailing whitespace in conversion of sound/core/hwdep.c
Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of mostly driver specific fixes, plus a maintainership
update for TI and a fix for DAPM driver removal paths.
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Merge tag 'asoc-fix-v5.11-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.11
A collection of mostly driver specific fixes, plus a maintainership
update for TI and a fix for DAPM driver removal paths.
FE is connected to two BEs, BE1 is active, BE2 is deactive.
When closing BE1, FE/BE1 is in HW_FREE state, then BE2 is
startup by mixer runtime update.
For FE is in HW_FREE state, dpcm_run_update_startup() will skip
BE2's startup because FE's state is HW_FREE, BE2 stays in FE's
be_clients list.
During FE's closed, the dpcm_fe_dai_close() will close all related
BEs, BE2 will be closed. This will lead to BE2's dpcm[stream].users
mismatch.
We need disconnet all pending BEs in the corner case.
Signed-off-by: zhucancan <zhucancan@vivo.com>
Link: https://lore.kernel.org/r/AAoArwDfDnoefyxzy2wyiaqm.1.1608885766936.Hmail.zhucancan@vivo.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove a stale comment about SSP0 being untested, the
bytcht_es8316 has supported SSP0 for a while now and this has
been successfully tested on a GP electrinic T701 tablet.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20210107120757.12051-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
* The HP ZBook Fury 15/17 G7 Mobile Workstation are using ALC285 codec
which is using 0x04 to control mute LED and 0x01 to control micmute LED.
* The right channel speaker is no sound and it needs to expose GPIO1 for
initialing AMP.
Add quirks to support them.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210106130549.100532-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here is a collection of USB- and HD-audio fixes: most of them are
device-specific quirks while one fix is for a regression by the
incorrect mutex unlock introduced in 5.11-rc1.
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Merge tag 'sound-5.11-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here is a collection of USB- and HD-audio fixes.
Most of them are device-specific quirks while one fix is for a
regression due to an incorrect mutex unlock introduced in this merge
window"
* tag 'sound-5.11-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/via: Fix runtime PM for Clevo W35xSS
ALSA: usb-audio: Add quirk for RC-505
ALSA: hda/hdmi: Fix incorrect mutex unlock in silent_stream_disable()
ALSA: hda/realtek: Enable mute and micmute LED on HP EliteBook 850 G7
ALSA: hda/realtek: Add two "Intel Reference board" SSID in the ALC256.
ALSA: hda/realtek: Add mute LED quirk for more HP laptops
ALSA: hda/conexant: add a new hda codec CX11970
ALSA: usb-audio: Add quirk for BOSS AD-10
ALSA: usb-audio: Fix UBSAN warnings for MIDI jacks
ALSA: hda/realtek - Modify Dell platform name
ALSA: hda/realtek - Fix speaker volume control on Lenovo C940
bclk_ratio is unused. Removes bclk_ratio and .set_bclk_ratio callback.
Removes snd_soc_dai_set_bclk_ratio() in a few machine drivers which are
obviously using rt1015.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20201224101854.3024823-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai