devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on
failure which should be checked with IS_ERR. Also use PTR_ERR for
returning error codes.
Reported-by: Takashi Iwai <tiwai@suse.de>
Fixes: a87a6653a2 ("ASoC: rt1308-sdw: add rt1308 SdW amplifier driver")
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200826163340.3249608-3-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on
failure which should be checked with IS_ERR. Also use PTR_ERR for
returning error codes.
Reported-by: Takashi Iwai <tiwai@suse.de>
Fixes: 56a5b7910e ("ASoC: codecs: max98373: add SoundWire support")
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200826163340.3249608-2-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This series includes fixes for error reporting, topology parsing and
runtime PM issues along with updates for DMIC support and IMX platforms.
Iulian Olaru (2):
ASoC: SOF: imx: Replace sdev->private with sdev->pdata->hw_pdata
ASoC: SOF: sof-of-dev: Add .arch_ops field
Jaska Uimonen (1):
ASoC: SOF: intel: hda: support also devices with 1 and 3 dmics
Keyon Jie (1):
ASoC: SOF: topology: fix the ipc_size calculation for process
component
Rander Wang (1):
ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
Ranjani Sridharan (2):
ASoC: SOF: Intel: hda: report error only for the last ROM init
iteration
ASoC: SOF: Intel: hda: add extended rom status dump to error log
sound/soc/sof/imx/Kconfig | 2 ++
sound/soc/sof/imx/imx8.c | 17 +++++++++----
sound/soc/sof/imx/imx8m.c | 10 +++++---
sound/soc/sof/intel/hda-codec.c | 4 +--
sound/soc/sof/intel/hda-loader.c | 42 +++++++++++++++++++-------------
sound/soc/sof/intel/hda.c | 26 +++++++++++++++++++-
sound/soc/sof/topology.c | 4 +--
7 files changed, 74 insertions(+), 31 deletions(-)
--
2.25.1
Add .arch_ops field in the sof_imx8x_ops structure.
The inclusion of this field will allow the usage of functions from
sof/core.c in order to print debug information such as the registers and
a stack dump in case of a firmware ops.
The SND_SOC_SOF_XTENSA is added in the imx/Kconfig file so the compilation
is successful.
Signed-off-by: Iulian Olaru <iulianolaru249@yahoo.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-8-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The correct way to save private data is to use sdev->pdata->hw_pdata.
Removed superfluous type-casts.
Signed-off-by: Iulian Olaru <iulianolaru249@yahoo.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-7-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Dump the extended ROM status information to the error logs
to aid with remote support. The analysis of these logs requires
access to non-public technical information.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-6-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The topology private struct is used for token parsing and its size
should not be included to the ipc_size, fix it here though it didn't
cause any real issue as the Firmware won't use this wrong-added data.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-5-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the dmic check code supports only devices with 2 or 4 dmics.
With other dmic counts the function will return 0. Lately we've seen
devices with only 1 dmic thus enable also configurations with 1, and
possibly 3, dmics. Add also topology postfix -1ch and -3ch for new dmic
configuration.
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-4-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When hda_codec_probe() doesn't initialize audio component, we disable
the codec and keep going. However,the resources are not released. The
child_count of SOF device is increased in snd_hdac_ext_bus_device_init
but is not decrease in error case, so SOF can't get suspended.
snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it
gets a error. Now copy this behavior to release resources and decrease
SOF device child_count to release SOF device.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The FW boot sequence includes multiple attempts for ROM init.
When it does take more than one attempt, we should not log the
errors encountered during the failed attempts and only log them
during the final iteration.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235040.1586478-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds an IPC initiated debug box region in the snd_sof_dev
structure, defined in soc/sof/sof-priv.h. It is initialized at loading,
in the sof_get_windows function from soc/sof/loader.c, in a similar manner
with the stream box and host box.
This region is useful because the firmware will put an error message
here so the kernel can read it in case of a dsp oops.
Signed-off-by: Iulian Olaru <iulianolaru249@yahoo.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235854.1588034-5-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The debug ABI can be extracted from the extended manifest content.
This information known at build time does not need to be provided
in a mailbox.
Signed-off-by: Karol Trzcinski <karolx.trzcinski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235854.1588034-4-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sdev->info_window is allocated with kmemdup and never freed, use devm_
version since this is only used for first boot.
Fixes: 8d809c15ac ('ASoC: SOF: ext_manifest: parse windows')
Cc: Karol Trzcinski <karolx.trzcinski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235854.1588034-3-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This step is needed to add possibility to pack sof_ipc_window inside
another one in used FW build tools - for example in extended manifest.
Structure reusability leads to easy parsing function reuse, so source
code is shorter and easier to maintain.
Using structures with constant size is less tricky and properly
supported by each toolchain by contrast to variable size elements.
This is minor ABI change - backward compatibility is kept.
Signed-off-by: Karol Trzcinski <karolx.trzcinski@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200825235854.1588034-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Galaxy Book Ion NT950XCJ-X716A (15 inches) uses the same ALC298
codec as other Samsung laptops which have the no headphone sound bug. I
confirmed on my own hardware that this fixes the bug.
This also correct the model name for the 13 inches version. It was
incorrectly referenced as NT950XCJ-X716A in commit e17f02d05. But it
should have been NP930XCJ-K01US.
Fixes: e17f02d055 ("ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Adrien Crivelli <adrien.crivelli@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200826084014.211217-1-adrien.crivelli@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add headphone gain and DAC filter controls, which use the same commands
as the AE-5. Also, change input source enumerated control item count to
exclude front microphone.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-20-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add AE-7 quirk data for setting of microphone. The AE-7 has no front
panel connector, so only rear-mic/line-in have new commands.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-19-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add structures containing the changes that need to happen on output
selection for each quirk. This should streamline the addition of new
quirks.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-9-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the output structures that were in use before and instead set the
DSP commands line by line. Now that the commands use is known, it makes
the functionality more clear this way.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-8-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the surround output selection and merge it with the speaker
output selection. Now that the extra commands that were being run on
surround output setting are known, there's no need to have it be
separate.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-7-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add bass redirection controls for surround outputs. This uses the DSP to
redirect audio below the bass redirection crossover frequency to the LFE
channel from the front/rear L/R speakers. This only goes into effect if
the speakers aren't set as full range, and only if the surround
configuration has an LFE channel.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-6-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add functions for setting full-range speakers and controls to
enable/disable the setting. Setting a speaker to full-range means that
the channels won't have their bass redirected to the LFE channel.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-5-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a surround channel configuration enumeration control. Setting up
different channel configurations allows the DSP to upmix stereo audio
into multi-channel audio, and allows for redirection of bass to a
subwoofer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-4-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cleanup the ca0132_mmio_init function, separating into two separate
functions, one for Sound Blaster Z/ZxR/Recon3D, and another for the
AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-2-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch extends support for DJM-250MK2 and allows recording.
However, DVS is not possible yet (see the comment in code).
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200825153113.6352-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ARRAY_SIZE() is the number of the elements but we want to use the
number of bytes. Fortunately, in this case the value is the same so it
doesn't affect runtime.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200825104623.GA278587@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
We use HDaudio and HDAudio, pick one to make searches easier.
No functionality change
Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Mirror change suggested in legacy HDaudio driver.
On SKL+ Intel platforms, the driver selection is handled by the
snd_intel_dspcfg, and when the HDaudio legacy driver is not selected,
be it with the auto-selection or user preferences with a kernel
parameter, the probe aborts with no logs, only a -ENODEV return value.
Having no dmesg trace, even with dynamic debug enabled, makes support
more complicated than it needs to be, and even experienced users can
be fooled. A simple dev_dbg() trace solves this problem.
BugLink: https://github.com/thesofproject/linux/issues/2330
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the probe relies on a workqueue, the completion is not signaled
by a return value. Mirror the log already present for PCI probe, so
that CI checks can test if the probe actually worked by filtering the
console logs.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When using dynamic debug, the console is swamped with verbose position
pointer logs, which really don't add much information. Move then to
vdbg to keep traces usable and allow for easier end-user support.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Virtual widgets are added to topology to be compatible with legacy
machine drivers. Reduce the log level for messages printed when
such widgets are ignored by the SOF driver.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like it was left over from the previous implementation of
DMIC PDM token parsing. It is not used anymore.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Several fields in struct snd_sof_dev are used as boolean flags, use
the "bool" type for them.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove two cases of redundant variable initialisation.
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200824200912.46852-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 059374fe9e ("ASoC: ti: merge .digital_mute() into .mute_stream()")
merged .digital_mute() into .mute_stream().
But it didn't rename ams_delta_digital_mute() to ams_delta_mute().
This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/87blizy5ts.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some sound card try to set 0 Hz as reset, but it is impossible.
This patch ignores it to avoid error return.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a6yjy5sy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 61eee4a7fc ("ALSA: hda: Add support for Loongson
7A1000 controller") to fix the following error on the Loongson LS7A
platform:
rcu: INFO: rcu_preempt self-detected stall on CPU
<SNIP>
NMI backtrace for cpu 0
CPU: 0 PID: 68 Comm: kworker/0:2 Not tainted 5.8.0+ #3
Hardware name: , BIOS
Workqueue: events azx_probe_work [snd_hda_intel]
<SNIP>
Call Trace:
[<ffffffff80211a64>] show_stack+0x9c/0x130
[<ffffffff8065a740>] dump_stack+0xb0/0xf0
[<ffffffff80665774>] nmi_cpu_backtrace+0x134/0x140
[<ffffffff80665910>] nmi_trigger_cpumask_backtrace+0x190/0x200
[<ffffffff802b1abc>] rcu_dump_cpu_stacks+0x12c/0x190
[<ffffffff802b08cc>] rcu_sched_clock_irq+0xa2c/0xfc8
[<ffffffff802b91d4>] update_process_times+0x2c/0xb8
[<ffffffff802cad80>] tick_sched_timer+0x40/0xb8
[<ffffffff802ba5f0>] __hrtimer_run_queues+0x118/0x1d0
[<ffffffff802bab74>] hrtimer_interrupt+0x12c/0x2d8
[<ffffffff8021547c>] c0_compare_interrupt+0x74/0xa0
[<ffffffff80296bd0>] __handle_irq_event_percpu+0xa8/0x198
[<ffffffff80296cf0>] handle_irq_event_percpu+0x30/0x90
[<ffffffff8029d958>] handle_percpu_irq+0x88/0xb8
[<ffffffff80296124>] generic_handle_irq+0x44/0x60
[<ffffffff80b3cfd0>] do_IRQ+0x18/0x28
[<ffffffff8067ace4>] plat_irq_dispatch+0x64/0x100
[<ffffffff80209a20>] handle_int+0x140/0x14c
[<ffffffff802402e8>] irq_exit+0xf8/0x100
Because AZX_DRIVER_GENERIC can not work well for Loongson LS7A HDA
controller, it needs some workarounds which are not merged into the
upstream kernel at this time, so it should revert this patch now.
Fixes: 61eee4a7fc ("ALSA: hda: Add support for Loongson 7A1000 controller")
Cc: <stable@vger.kernel.org> # 5.9-rc1+
Signed-off-by: Tiezhu Yang <yangtiezhu@loongson.cn>
Link: https://lore.kernel.org/r/1598348388-2518-1-git-send-email-yangtiezhu@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WAKEEN bits are used to indicate which bits in the
STATESTS register may cause wake event during the codec
state change request. Configure the WAKEEN register for
the Tegra to detect the wake events.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Tegra HDA codec HW implementation has an issue related to not
swapping the 2 channel Audio Sample Packet(ASP) channel mapping.
Whatever the FL and FR mapping specified the left channel always
comes out of left speaker and right channel on right speaker. So
add condition to disallow the swapping of FL,FR during the playback.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ca0106_spi_write() returns 1 on error, snd_ca0106_pcm_power_dac()
is returning the error code directly, and the caller is expecting an
negative error code
Signed-off-by: Tong Zhang <ztong0001@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200824224541.1260307-1-ztong0001@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the playback & capture streams are stopped simultaneously, the
SOF PCI device will remain pm_runtime active. The root-cause is a race
condition with two threads reaching the trigger function at the same
time. They see another stream is active so the dapm pin is not
disabled, so the codec remains active as well as the parent PCI
device.
For max98373, the capture stream provides feedback when playback is
working and it is unused when playback is stopped. So the dapm pin
should be set only when playback is active.
Fixes: 94d2d08974 ('ASoC: Intel: Boards: tgl_max98373: add dai_trigger function')
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This series updates the tables used to select SoundWire configurations
for CometLake and TigerLake, and adds support for SDCA (SoundWire
Device Class for Audio) codecs in the common machine driver. These
codec drivers are still being tested on early silicon/boards and will
be contributed at a later time.
For TigerLake Chromebooks a new DMI quirk is added, as well as a means
to override the topology names. A pm_runtime fix is also provided to
deal with playback/capture dependencies with an amplifier w/
feedback. I also included a minor codec correction for the TGL
amplifier.
Bard Liao (5):
ASoC: Intel: modify SoundWire version id in acpi match table
ASoC: Intel: sof_sdw: check SoundWire version when matching codec
ASoC: Intel: sof_sdw: rename id as part_id
ASoC: Intel: sof_sdw: add rt711 rt1316 rt714 SDCA codec support.
ASoC: Intel: sof_sdw: clean-up inclusion of header files
Pierre-Louis Bossart (5):
ASoC: Intel: soc-acpi: cnl: add support for rt5682 on SoundWire link2
ASoC: Intel: sof-soundwire: add support for rt5682 on link2
ASoC: Intel: soc-acpi: mirror CML and TGL configurations
ASoC: Intel: soc-acpi: add support for SDCA boards
ASoC: codecs: max98373-sdw: add missing test on resume
Rander Wang (2):
ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case
ASoC: Intel: sof_sdw: Add support for product Ripto
Sathyanarayana Nujella (2):
ASoC: Intel: sof_rt5682: override quirk data for tgl_max98373_rt5682
ASoC: SOF: Add topology filename override based on dmi data match
sound/soc/codecs/max98373-sdw.c | 3 +
sound/soc/intel/boards/Kconfig | 3 +
sound/soc/intel/boards/Makefile | 7 +-
sound/soc/intel/boards/sof_maxim_common.c | 7 +-
sound/soc/intel/boards/sof_rt5682.c | 13 ++
sound/soc/intel/boards/sof_sdw.c | 98 +++++++---
sound/soc/intel/boards/sof_sdw_common.h | 22 ++-
sound/soc/intel/boards/sof_sdw_dmic.c | 1 +
sound/soc/intel/boards/sof_sdw_max98373.c | 2 +
sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +
sound/soc/intel/boards/sof_sdw_rt1316.c | 113 ++++++++++++
sound/soc/intel/boards/sof_sdw_rt5682.c | 2 +
sound/soc/intel/boards/sof_sdw_rt700.c | 2 +
sound/soc/intel/boards/sof_sdw_rt711.c | 2 +
sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 174 ++++++++++++++++++
sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 42 +++++
.../intel/common/soc-acpi-intel-cml-match.c | 79 +++++++-
.../intel/common/soc-acpi-intel-cnl-match.c | 33 +++-
.../intel/common/soc-acpi-intel-icl-match.c | 10 +-
.../intel/common/soc-acpi-intel-tgl-match.c | 165 ++++++++++++++++-
sound/soc/sof/intel/hda.c | 8 +-
sound/soc/sof/sof-pci-dev.c | 24 +++
22 files changed, 764 insertions(+), 48 deletions(-)
create mode 100644 sound/soc/intel/boards/sof_sdw_rt1316.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt711_sdca.c
create mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c
base-commit: fcea8b023a
--
2.25.1
Regmap initialization may return -EPROBE_DEFER for clock
may not be ready, so check -EPROBE_DEFER error type before
start another Regmap initialization.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1598255887-1391-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
"AVDD" is for analog power supply, "DVDD" is for digital power
supply, they can improve the power management.
As the regulator is enabled in pm runtime resume, which is
behind the component driver probe, so accessing registers in
component driver probe will fail. Fix this issue by enabling
regcache_cache_only after pm_runtime_enable.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1598190877-9213-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
"struct snd_soc_dapm_widget" and "struct snd_kcontrol_new" are used in most
of these .c files. Adding the header files to prevent from depending on
<sound/soc.h>
Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add topology filename override based on system DMI data matching,
typically to account for a different hardware layout.
In ACPI based systems, the tplg_filename is pre-defined in an ACPI
machine table. When a DMI quirk is detected, the
sof_pdata->tplg_filename is not set with the hard-coded ACPI value,
and instead is set with the DMI-specific filename.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A Chrome System based on tgl_max98373_rt5682 has different SSP interface
configurations. Using DMI data of this variant DUT, override quirk
data.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Ripto is another product based on TGL with the same
audio hardware configuration as Volteer.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "id" field in sof_sdw_codec_info struct is actually the "part
id". Rename to prevent confusions.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some codecs with the same part id but different SoundWire versions
have different configurations. So we have to separate them in
codec_info_list[].
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All existing SoundWire codecs follow the same pattern on resume,
except for this codec which doesn't test if the hardware is
initialized.
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the playback & capture streams are stopped simultaneously, the
SOF PCI device will remain pm_runtime active. The root-cause is a race
condition with two threads reaching the trigger function at the same
time. They see another stream is active so the dapm pin is not
disabled, so the codec remains active as well as the parent PCI
device.
For max98373, the capture stream provides feedback when playback is
working and it is unused when playback is stopped. So the dapm pin
should be set only when playback is active.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The description and board layout is similar to previous ones for
CometLake and TigerLake, except for a bump to SoundWire 1.2 and
updates to part numbers to reflect the SDCA (SoundWire Device Class
for Audio) hardware support.
Note that one of the RT1316 amplifiers uses a non-zero UniqueID which
is not required and will be ignored.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some TGL devices use the same audio hardware as on CML platforms, with
RT711 on link0, RT1308 on link1 and optionally link2, and RT715 on
link 3.
To clarify configurations, the rt1308 configurations are split between
single amp on link1 and dual amps on link1. The case with two amps on
different links is already identified with the group1 attribute.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The UpExtreme board provides support for SoundWire link2 in 2 of the 3
advanced modes. Let's use it w/ rt5682.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one of the configurations for rt5682 w/ the Up Extreme Advanced
Audio mode using the SoundWire link2.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SoundWire version id of the existing RT1308, RT711, and RT715
codecs should be 2 (index for SoundWire 1.1), it was mistakenly set as
1 which pointed to the wrong version (SoundWire 1.0).
This off-by-one error had no functional impact so far since the
version number was not used, however in future patches this version
will be required.
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200821195603.215535-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 51ab5d77dc ("ALSA: usb-audio: Properly match with audio
interface class") converted the quirk entries that have both vid/pid
pair and bInterface fields to match with all those with a new macro
USB_AUDIO_CLASS(). However, it turned out that those are false
conversions; all those (but the unknown KeithMcMillen device) are
actually with vendor-specific interface class, hence the conversions
broke the matching.
This patch corrects those entries to the right one,
USB_DEVICE_VENDOR_SPEC() (and USB_DEVICE() for KeithMcMillen to be
sure), and drop the unused USB_AUDIO_CLASS macro again.
Fixes: 51ab5d77dc ("ALSA: usb-audio: Properly match with audio interface class")
Reported-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200823113251.10175-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct quirk table entries for Lenovo ThinkStation P620, too.
The name and profile strings are now set from a different table, hence
removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If USB autosuspend is enabled, both front and rear panel can no longer
detect jack insertion.
Enable USB remote wakeup, i.e. needs_remote_wakeup = 1, doesn't help
either.
So disable USB autosuspend to prevent missing jack detection event.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200823105854.26950-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tascam FE-8 is known to support communication by asynchronous transaction
only. The support can be implemented in userspace application and
snd-firewire-ctl-services project has the support. However, ALSA
firewire-tascam driver is bound to the model.
This commit changes device entries so that the model is excluded. In a
commit 53b3ffee78 ("ALSA: firewire-tascam: change device probing
processing"), I addressed to the concern that version field in
configuration differs depending on installed firmware. However, as long
as I checked, the version number is fixed. It's safe to return version
number back to modalias.
Fixes: 53b3ffee78 ("ALSA: firewire-tascam: change device probing processing")
Cc: <stable@vger.kernel.org> # 4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200823075537.56255-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes over several drivers, but all are driver-
specific and nothing looks scary. Slightly large changes are seen in
ASoC qcom driver for the bugs that were revealed by the recent ASoC
core change to report the invalid register access errors. Also ASoC
fsl got a slight intensive change for the distortion fix. Others are
only trivial fixes or device-specific quirks.
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Merge tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes over several drivers, but all are driver-
specific and nothing looks scary.
Slightly large changes are seen in ASoC qcom driver for the bugs that
were revealed by the recent ASoC core change to report the invalid
register access errors. Also ASoC fsl got a slight intensive change
for the distortion fix.
Others are only trivial fixes or device-specific quirks"
* tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
ALSA: hda: avoid reset of sdo_limit
ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
ALSA: usb-audio: ignore broken processing/extension unit
ASoC: intel: Fix memleak in sst_media_open
ASoC: wm8994: Avoid attempts to read unreadable registers
ASoC: msm8916-wcd-analog: fix register Interrupt offset
ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
ALSA: hda/realtek: Add model alc298-samsung-headphone
ALSA: usb-audio: Update documentation comment for MS2109 quirk
ALSA: isa: fix spelling mistakes in the comments
ALSA: usb-audio: Add capture support for Saffire 6 (USB 1.1)
ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
ASoC: q6routing: add dummy register read/write function
ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM
ASoC: Make soc_component_read() returning an error code again
ASoC: amd: Replacing component->name with codec_dai->name.
ASoC: fsl: Fix unused variable warning
ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n
ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n
ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n
...
Previous improvements around handling device and codec level
probe functionality added the possibility of the voltage level
being undefined for the scenario where the IO voltage retrieved
from the regulator supply was below 1.2V, whereas previously the
code defaulted to the 2.5V to 3.6V range in that case. This
commit restores the default value to avoid this happening.
Fixes: aa5b18d1c2 ("ASoC: da7219: Move soft reset handling to codec level probe")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/20200821142259.C2ECE3FB96@swsrvapps-01.diasemi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On Linux 5.9-rc1 I get the following warning with apq8016-sbc:
WARNING: CPU: 2 PID: 69 at sound/core/init.c:207 snd_card_new+0x36c/0x3b0 [snd]
CPU: 2 PID: 69 Comm: kworker/2:1 Not tainted 5.9.0-rc1 #1
Workqueue: events deferred_probe_work_func
pc : snd_card_new+0x36c/0x3b0 [snd]
lr : snd_card_new+0xf4/0x3b0 [snd]
Call trace:
snd_card_new+0x36c/0x3b0 [snd]
snd_soc_bind_card+0x340/0x9a0 [snd_soc_core]
snd_soc_register_card+0xf4/0x110 [snd_soc_core]
devm_snd_soc_register_card+0x44/0xa0 [snd_soc_core]
apq8016_sbc_platform_probe+0x11c/0x140 [snd_soc_apq8016_sbc]
This warning was introduced in
commit 81033c6b58 ("ALSA: core: Warn on empty module").
It looks like we are supposed to set card->owner to THIS_MODULE.
Fix this for all the qcom ASoC drivers.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 79119c7986 ("ASoC: qcom: Add Storm machine driver")
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Fixes: a6f933f63f ("ASoC: qcom: apq8096: Add db820c machine driver")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200820154511.203072-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
We may allocate some resources in sof_sdw_codec_info .init function.
Adding a corresponding .exit function can help to release these resources.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200820134542.8682-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The TLV320AIC32x4 is commonly used on TQ-Systems starterkit mainboards
for i.MX-based SoMs (i.MX6Q/DL, i.MX6UL, i.MX7) and LS1021A.
Signed-off-by: Matthias Schiffer <matthias.schiffer@ew.tq-group.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200821071153.7317-2-matthias.schiffer@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use constraint to make sure the period size could always be multiple
of 1ms to align with the fundamental design/limitation of firmware.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/1596198365-10105-2-git-send-email-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Select SoundWire capabilities on newer Intel platforms, starting with
CannonLake/CoffeeLake/CometLake.
As done for HDaudio, the SoundWire link is an opt-in capability. We
explicitly test for ACPI to avoid warnings on unmet dependencies on
the SoundWire side.
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/20200819124429.3785-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently drvdata->clks is not being checked for an allocation failure,
leading to potential null pointer dereferencing. Fix this by adding a
check and returning -ENOMEM if an error occurred.
Fixes: 1220f6a76e ("ASoC: qcom: Add common array to initialize soc based core clocks")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Rohit kumar <rohitkr@codeaurora.org>
Addresses-Coverity: ("Dereference null return value")
Link: https://lore.kernel.org/r/20200819160103.164893-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The s3c_gpio_cfgall_range() function is an internal interface of the
samsung gpio driver and should not be called directly by drivers, so
move the iis pin initialization into the boards.
This means the pin configuration is only run once at early boot, rather
than each time the driver binds, but the effect should be the same.
Note that the s3c2412-i2s driver has no boards using it in mainline linux,
the driver gets selected for the jive machine but is never instantiated.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-28-krzk@kernel.org
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
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Merge tag 'samsung-platdrv-boards' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into next/soc-s3c-cleanup
Pull Samsung S3C ASoC cleanup patches from Mark Brown. These patches
are part of the entire cleanup series so all further work depends on
them.
There are a few entries in the quirk table that set the device ID with
USB_DEVICE() macro while having an extra bInterfaceClass field. But
bInterfaceClass field is never checked unless the proper match_flags
is set, so those may match incorrectly with all interfaces.
Introduce another macro to match with the vid/pid pair and the audio
class interface, and apply it to such entries, so that they can match
properly.
Link: https://lore.kernel.org/r/20200817082140.20232-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new macro USB_AUDIO_DEVICE() for the entries matching with
the pid/vid pair and the class/subclass, and remove the open-code.
Link: https://lore.kernel.org/r/20200817082140.20232-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we've added the devices that need vendor/product string renames
or the profile setup into the standard quirk table in quirks-table.h.
This table is imported into the primary USB audio device entry, hence
it's all exported for the probing so that udev and co can take a look
at it. OTOH, for renaming or profile setup, we don't need to expose
those explicit entries because the probe itself follows the standard
way. That said, we're exposing unnecessarily too many entries.
This patch moves such internal quirk entries into the own table, and
reduces the exported device table size. Along with the moving items,
re-arrange the entries in the proper order.
Link: https://lore.kernel.org/r/20200817082140.20232-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The plat-samsung directory and mach-s5pv210 can be build
completely independently, so split the two Kconfig symbols
CONFIG_PLAT_SAMSUNG and CONFIG_ARCH_S5PV210.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Thierry Reding <thierry.reding@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-18-krzk@kernel.org
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Due to a mistake made while reordering patches, commit 90cac93297
("ASoC: sun8i-codec: Fix DAPM to match the hardware topology") added
the sun8i_codec_component_probe function without referencing it from
the component definition. Add the reference so the probe function gets
called as expected.
Fixes: 90cac93297 ("ASoC: sun8i-codec: Fix DAPM to match the hardware topology")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200819034038.46418-1-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The constant requires indirectly including a machine header file,
but it's not actually used any more since commit 87b132bc03 ("ASoC:
samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so
remove it completely.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-27-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Avoid machine specific headers by using a gpio lookup table
combined with a platform_driver for this board.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-26-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Avoid machine specific headers by using a gpio lookup table
combined with a platform_driver for this board.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-25-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Avoid machine specific headers by using a gpio lookup table
combined with a platform_driver for this board.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200806182059.2431-24-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
By default 'sdo_limit' is initialized with a default value of '8'
as per spec. This is overridden in cases where a different value is
required. However this is getting reset when snd_hdac_bus_init_chip()
is called again, which happens during runtime PM cycle.
Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()'
function which would be called only once.
Fixes: 67ae482a59 ("ALSA: hda: add member to store ratio for stripe control")
Cc: <stable@vger.kernel.org>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1597851130-6765-1-git-send-email-spujar@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add version and class information explicitly to prepare for support
for new devices.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200818141435.29205-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Different modules for HDMI codec are used depending on the
"hda_codec_use_common_hdmi" option being enabled or not. Driver private
context for both of them is different.
This leads to null-pointer dereference error when driver tries to set
autosuspend delay for HDMI codec while the option is off (hdac_hdmi
module is used for HDMI).
Change the string in conditional statement to "ehdaudio0D0" to ensure
that only the HDAudio codec is handled by this function.
Fixes: 5bf73b1b1d ("ASoC: intel/skl/hda - fix oops on systems without i915 audio codec")
Signed-off-by: Mateusz Gorski <mateusz.gorski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200722173524.30161-1-mateusz.gorski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A bunch of fixes that came in during the merge window, mostly for issues
that were uncovered by the changes to report errors on invalid register
access plus one important fix in that code itself.
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Merge tag 'asoc-fix-v5.9-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.9
A bunch of fixes that came in during the merge window, mostly for issues
that were uncovered by the changes to report errors on invalid register
access plus one important fix in that code itself.
It seems the datasheet has never used the word slave for this error
status bit and has always used the term address error. So update the
driver to match the datasheets and also in the process align a bit
better with avoiding the use of such words where possible.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200818160126.4852-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patchset adds gapless compressed audio support on q6asm.
Gapless on q6asm is implemented using 2 streams in a single q6asm session.
First few patches such as stream id per each command, gapless flags
and silence meta data are for preparedness for adding gapless support.
Last patch implements copy callback to allow finer control over buffer offsets,
specially in partial drain cases.
This patchset is tested on RB3 aka DB845c platform.
This patchset as it is will support gapless however QDSP can also
support switching decoders on a single stream. Patches to support such feature
are send in different patchset which involves adding generic interfaces.
Thanks,
srini
Changes since v2:(mostly suggested by Pierre)
- removed unnessary kernel style comments,
- moved TIMESTAMP flag to respective patch.
- move preparatory code from gapless support patch to new one.
- fix subject prefix of one patch.
- add comments to clarify valid stream_ids
Srinivas Kandagatla (10):
ASoC: q6asm: rename misleading session id variable
ASoC: q6asm: make commands specific to streams
ASoC: q6asm: use flags directly from q6asm-dai
ASoC: q6asm: add length to write command token
ASoC: q6asm: add support to remove intial and trailing silence
ASoC: q6asm: add support to gapless flag in q6asm open
ASoC: q6asm-dai: add next track metadata support
ASoC: q6asm-dai: prepare set params to accept profile change
ASoC: q6asm-dai: add gapless support
ASoC: q6asm-dai: add support to copy callback
sound/soc/qcom/qdsp6/q6asm-dai.c | 414 +++++++++++++++++++++++--------
sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++----
sound/soc/qcom/qdsp6/q6asm.h | 49 ++--
3 files changed, 469 insertions(+), 163 deletions(-)
--
2.21.0
This patch set reorganises and fixes device and codec level probe/remove
handling within the driver, to allow clean probe and remove at the codec level.
This set relates to an issue raised by Yong Zhi where a codec level re-probe
would fail due to clks still being registered from the previous instantiation.
In addition some improvements around regulator handling and soft reset have
also been included.
Adam Thomson (3):
ASoC: da7219: Move required devm_* allocations to device level code
ASoC: da7219: Move soft reset handling to codec level probe
ASoC: da7219: Fix clock handling around codec level probe
sound/soc/codecs/da7219-aad.c | 85 +++++---
sound/soc/codecs/da7219-aad.h | 3 +
sound/soc/codecs/da7219.c | 493 +++++++++++++++++++++++-------------------
sound/soc/codecs/da7219.h | 1 +
4 files changed, 328 insertions(+), 254 deletions(-)
--
1.9.1
This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing
v2: Fixed the build error from snd_soc_component_read32
v1: initial ASoC: codec: tlv3204: Codec workaround series
Michael Sit Wei Hong (3):
ASoC: codec: tlv3204: Enable 24 bit audio support
ASoC: codec: tlv3204: Increased maximum supported channels
ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset
sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++---------
1 file changed, 44 insertions(+), 16 deletions(-)
--
2.17.1
This series performs some minor cleanup on the driver for the analog
codec in the Allwinner A64, and hooks up the existing mute switches to
DAPM widgets, in order to provide improved power management.
Changes since v1:
- Collected Acked-by/Reviewed-by tags
- Used SOC_MIXER_NAMED_CTL_ARRAY to avoid naming a widget "Earpiece"
Samuel Holland (8):
ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits
ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend
ASoC: sun50i-codec-analog: Group and sort mixer routes
ASoC: sun50i-codec-analog: Make headphone routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for headphone switch
ASoC: sun50i-codec-analog: Make line out routes stereo
ASoC: sun50i-codec-analog: Enable DAPM for line out switch
ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch
sound/soc/sunxi/sun50i-codec-analog.c | 176 ++++++++++++++++----------
1 file changed, 111 insertions(+), 65 deletions(-)
--
2.26.2
This series fixes a couple of issues with the digital audio codec in the
Allwinner A64 SoC:
1) Left/right channels were swapped when playing/recording audio
2) DAPM topology was wrong, breaking some kcontrols
This is the minimum set of changes necessary to fix these issues in a
backward-compatible way. For that reason, some DAPM widgets still have
incorrect or confusing names; those and other issues will be fixed in
later patch sets.
Samuel Holland (7):
ASoC: dt-bindings: Add a new compatible for the A64 codec
ASoC: sun8i-codec: Fix DAPM to match the hardware topology
ASoC: sun8i-codec: Add missing mixer routes
ASoC: sun8i-codec: Add a quirk for LRCK inversion
ARM: dts: sun8i: a33: Update codec widget names
arm64: dts: allwinner: a64: Update codec widget names
arm64: dts: allwinner: a64: Update the audio codec compatible
.../sound/allwinner,sun8i-a33-codec.yaml | 6 +-
arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 4 +-
arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-
.../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +-
.../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +-
.../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +-
.../dts/allwinner/sun50i-a64-pinebook.dts | 8 +-
.../dts/allwinner/sun50i-a64-pinephone.dtsi | 8 +-
.../boot/dts/allwinner/sun50i-a64-pinetab.dts | 8 +-
.../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +-
.../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +-
arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 11 +-
sound/soc/sunxi/sun8i-codec.c | 137 ++++++++++++++----
13 files changed, 155 insertions(+), 71 deletions(-)
--
2.26.2
This patch series drops a printk message down to dev_dbg() because it
was noisy and then migrates this driver to use clk_hw based APIs instead
of clk based APIs because this device is a clk provider, not a clk
consumer. I've only lightly tested the last two patches but I don't have
all combinations of clks for this device.
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Stephen Boyd (3):
ASoC: rt5682: Use dev_dbg() in rt5682_clk_check()
ASoC: rt5682: Drop usage of __clk_get_name()
ASoC: rt5682: Use clk_hw based APIs for registration
sound/soc/codecs/rt5682.c | 73 ++++++++++++---------------------------
sound/soc/codecs/rt5682.h | 2 --
2 files changed, 23 insertions(+), 52 deletions(-)
Based on the last patch to this driver in linux-next.
base-commit: 6301adf942
--
Sent by a computer, using git, on the internet
On some platform(.e.g. i.MX8QM MEK), the "extal" clock is different
with the mclk of codec, then the clock rate is also different.
So it is better to get clock rate of "extal" rate by clk_get_rate,
don't reuse the clock rate of mclk.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1597047103-6863-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently list of Qualcomm drivers is growing, so put them in to a
proper menu so that it does not mix up with other ASOC configs in menuconfig.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811105818.7890-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This allows solutions like ALSA UCM to utilize hardware mono downmix
for cases where mono output to a single speaker is desired only in
specific situations (like on a mobile phone).
Signed-off-by: Sebastian Krzyszkowiak <sebastian.krzyszkowiak@puri.sm>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/3662154.EqNIRYjrc8@pliszka
Signed-off-by: Mark Brown <broonie@kernel.org>
There are a couple of occurrences of "the the" in the Kconfig
text. Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Michal Simek <michal.simek@xilinx.com>
Link: https://lore.kernel.org/r/20200817224706.6139-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the repeated words {that, the} in comments.
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200808012156.10827-1-rdunlap@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow recommendation in Documentation/scheduler/completion.rst and
use macro to declare local 'struct completion'
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200813175442.59067-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new common snd_soc_of_parse_aux_devs() helper function
to parse auxiliary devices from the device tree. The new helper
is just a copy of meson_card_add_aux_devices() so there is no
functional change.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200801100257.22658-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new common snd_soc_of_parse_aux_devs() helper function
to parse auxiliary devices from the device tree. The code is slightly
different but the binding that is parsed is exactly the same.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200801100257.22658-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card.c and meson-card-utils.c use pretty much the same
helper function to parse auxiliary devices from the device tree.
Make it easier for other drivers to parse these from the device tree
as well by adding a shared helper function to soc-core.c.
snd_soc_of_parse_aux_devs() is pretty much a copy of
meson_card_add_aux_devices() from meson-card-utils.c
with two minor changes:
- Make property name configurable as parameter
- Change dev_err() message slightly for consistency with other
error messages in soc-core.c
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200801100257.22658-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
"AVDD" is for analog power supply, "DVDD" is for digital power
supply, they can improve the power management.
As the regulator is enabled in pm runtime resume, which is
behind the component driver probe, so accessing registers in
component driver probe will fail. Fix this issue by enabling
regcache_cache_only after pm_runtime_enable.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1597397561-2426-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add an binary mixer 'ELD' to each HDMI PCM device so user space
could read the ELD data of external HDMI display.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Link: https://lore.kernel.org/r/20200818004413.12852-1-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/skylake/skl-topology.c:2879:29: style: Variable
'block_size' is assigned a value that is never used. [unreadVariable]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cppcheck complains about possible NULL pointer dereferences but the
assignments are actually not needed before walking through lists.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-20-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/skylake/skl-sst-utils.c:240:10: style: Variable 'ret'
is assigned a value that is never used. [unreadVariable]
int ret = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/skylake/skl-sst-cldma.c:248:10: style: Variable 'ret'
is assigned a value that is never used. [unreadVariable]
int ret = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/skylake/skl-nhlt.c:203:21: style: Variable 'rate' is
assigned a value that is never used. [unreadVariable]
unsigned long rate = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/haswell/sst-haswell-ipc.c:430:8: style: Variable 'i'
is assigned a value that is never used. [unreadVariable]
sound/soc/intel/haswell/sst-haswell-ipc.c:1792:8: style: Variable 'id'
is assigned a value that is never used. [unreadVariable]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-16-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/atom/sst/sst_loader.c:401:43: style: Redundant
condition: If 'EXPR == 4', the comparison 'EXPR != 3' is always
true. [redundantCondition]
if (sst_drv_ctx->sst_state != SST_RESET ||
^
In this case, if sst_state == SST_SHUTDOWN then the first test is
already true. 2014 bug, yay.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/atom/sst/sst.c:52:20: style: Variable 'size' is
assigned a value that is never used. [unreadVariable]
unsigned int size = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning
return ret_val;
^
sound/soc/intel/atom/sst-mfld-platform-pcm.c:384:6: note: If condition 'ret_val' is true, the function will return/exit
if (ret_val)
^
sound/soc/intel/atom/sst-mfld-platform-pcm.c:387:9: note: Returning identical expression 'ret_val'
return ret_val;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/atom/sst/sst.c:427:13: style: Variable 'ret' is
assigned a value that is never used. [unreadVariable]
int i, ret = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/atom/sst/sst.c:373:2: warning: Assignment of function
parameter has no effect outside the function. Did you forget
dereferencing it? [uselessAssignmentPtrArg]
ctx = NULL;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
sound/soc/intel/atom/sst-mfld-platform-compress.c:46:14: style:
Variable 'ret_val' is assigned a value that is never
used. [unreadVariable]
int ret_val = 0;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck complains of a possible NULL pointer dereference but setting
a pointer before using list_for_each_entry() is not useful.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813200147.61990-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices have broken extension unit where getting current value
doesn't work. Attempt that once when creating mixer control for it. If
it fails, just ignore it, so that it won't cripple the device entirely
(and/or make the error floods).
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously the driver would use devm_* related functions at
the codec level probe() to allocate clock resources for MCLK
and the DAI clocks exposed by the device. This caused issues
when registering clocks on a re-probe (no device level
remove/prove involved) as the devm_* resources were never
freed up so the clocks were still registered from the previous
codec level probe().
This commit updates the clock handling for MCLK usage and DAI
clock provision to fix this discrepancy and allow the codec level
probe/remove functionality to operate as intended.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/b92c461baeed27a6cd92e59e36a55c2547218683.1597164865.git.Adam.Thomson.Opensource@diasemi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As part of the reorganisation of the device level and codec
level probe functionlity, the soft reset handling should really
reside at the codec level and after the instantiation of supplies.
This commit makes the relevant changes to support this change of
scope including the remove of devm_* functions being called for
regulator instantiation at the codec level.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/f7603a4855647429b754ce76f887ec441622015c.1597164865.git.Adam.Thomson.Opensource@diasemi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During gapless playback, its possible for previous track to
end at unaligned boundary, starting next track on the same
boundary can lead to unaligned address exception in dsp.
So implement copy callback for finer control on the buffer offsets.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-11-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to gapless playback by implementing metadata,
next_track, drain and partial drain support.
Gapless on Q6ASM is implemented by opening 2 streams in a single
q6asm stream and toggling them on next track.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-10-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
rearrange code so that it will be easy to change the codec
profile at runtime. This means moving exiting set_params
to an internal wrapper which can be called when codec
profile changes.
This is also preparing the code for easy to use in gapless cases.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-9-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to metadata required to do a gapless playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-8-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to gapless flag to q6asm_open_write().
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-7-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to ASM_DATA_CMD_REMOVE_INITIAL_SILENCE
and ASM_DATA_CMD_REMOVE_TRAILING_SILENCE q6asm command to support
compressed metadata for gapless playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-6-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Add length to write command packet token so that we can track exactly
how many bytes are consumed by DSP in the command reply.
This is useful in some use-cases where the end of the file/stream
is not aligned with period size.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
use flags set by q6asm-dais directly!
This will be useful gapless case where write needs a special flag to indicate
that last buffer.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Each ASM session can have multiple streams attached to it,
current design was to allow only one static stream id 1 per each session.
However for use-case like gapless, we would need 2 streams to open per session.
This patch converts all the q6asm apis to take stream id as argument
to allow multiple streams to open on a single session, This is useful
for gapless playback cases.
Now the dai driver can specify which stream id for each command.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Each q6asm session can have multiple streams, mixing usage of these
names in variable are bit misleading to reader, so rename them accordingly.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200727093806.17089-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable I2S TDM audio capture for Intel Keem Bay platform.
The I2S TDM will support 4 channel and 8 channel audio capture only.
4 channel and 8 channel audio capture operates only in slave mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200811041836.999-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Moving GPIO reset to a later stage and before clock registration to
ensure that the host system and codec clocks are in sync. If the host
register clock values prior to gpio reset, the last configured codec clock
is registered to the host. The codec then gets gpio resetted setting the
codec clocks to their default value, causing a mismatch. Host system will
skip clock setting thinking the codec clocks are already at the requested
rate.
ADC reset is added to ensure the next audio capture does not have
undesired artifacts. It is probably related to the original code
where the probe function resets the ADC prior to 1st record.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-4-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Increased maximum supported channel to 8 channels for audio capture
running in TDM mode.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-3-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable 24 bit in 32 bit container audio support.
Using the params_physical_width to differentiate
24 bit in 32 bit container and 24 bit in 24 bit container modes.
Use the sample rate, bit depth and channel parameters to
calculate the bit clock needed.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200812094631.4698-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the earpiece mute switch in the DAPM graph, both the
earpiece amplifier and the Mixer/DAC inputs can be powered off when
the earpiece is muted.
While the widget is really just a simple switch, it is represented
as a "mixer with named controls" to avoid including the widget name
in the kcontrol name. Otherwise, it is not possible to give the widget
an accurate, descriptive name without changing the kcontrol name
seen by userspace (which should be stable).
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726025334.59931-9-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the line out mute switch in the DAPM graph, the
Mixer/DAC inputs can be powered off when the line output is muted.
The line outputs have an unusual routing scheme. The left side mute
switch is between the source selection and the amplifier, as usual.
The right side source selection comes *after* its amplifier (and
after the left side amplifier), and its mute switch controls
whichever source is currently selected. This matches the diagram in
the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-8-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This matches the hardware more accurately, and is necessary for
including the (stereo) line out mute switch in the DAPM graph.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-7-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
By including the headphone mute switch to the DAPM graph, both the
headphone amplifier and the Mixer/DAC inputs can be powered off when
the headphones are muted.
The mute switch is between the source selection and the amplifier,
as per the diagram in the SoC manual.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-6-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This matches the hardware more accurately, and is necessary for
including the (stereo) headphone mute switch in the DAPM graph.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-5-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Sort the controls in the same order as the bits in the register. Then
group the routes by sink, and sort them in the same order as the
controls. This makes it much easier to verify that all mixer inputs are
accounted for.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The clock must be running for the zero-crossing mute functionality.
However, it must be gated for VDD-SYS to be turned off during system
suspend. Disable it in the suspend callback, after everything has
already been muted, to avoid pops when muting/unmuting outputs.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-3-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The same enable bits are currently used for both the "Left/Right ADC"
and the "Left/Right ADC Mixer" widgets. This happens to work in practice
because the widgets are always enabled/disabled at the same time, but
each register bit should only be associated with a single widget.
To keep symmetry with the DAC widgets, keep the bits on the ADC widgets,
and remove them from the ADC Mixer widgets.
Fixes: 42371f327d ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Reported-by: Ondrej Jirman <megous@megous.com>
Signed-off-by: Samuel Holland <samuel@sholland.org>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Link: https://lore.kernel.org/r/20200726025334.59931-2-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warnings:
sound/soc/intel/boards/bdw-rt5650.c:91:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
sound/soc/intel/boards/bdw-rt5677.c:144:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
sound/soc/intel/boards/broadwell.c:91:23: style: Local variable
'channels' shadows outer variable [shadowVariable]
This was fixed earlier in other machine drivers but keeps coming back
with copy/paste.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813175839.59422-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cppcheck reports the following warning:
sound/soc/sof/intel/hda-codec.c:191:1: style: Label 'error' is not
used. [unusedLabel]
This label is indeed only used conditionally, move it where it's
actually used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813175839.59422-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the A64, as tested using the PinePhone, the current code causes the
left/right channels to be swapped during I2S playback from the CPU on
AIF1, and breaks DSP_A communication with the modem on AIF2. Both of
these are fixed when LRCK is no longer inverted.
Trusting that the comment in the code is correct, the existing behavior
is kept for the A33.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-5-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun8i-codec driver provides ALSA controls for enabling/disabling
each of the inputs to the AIF1 Slot 0 and DAC mixers. For two of these
inputs (ADC->DAC and AIF1 DA0->AIF1 AD0), the audio source is
implemented, so the mixer inputs can be used.
However, because the DAPM routes are missing, these mixer inputs only
work when both the source and the mixer happen to be part of other
active audio paths. Adding the appropriate routes makes these ALSA
controls function all of the time.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-4-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The A33/A64 digital codec has 4 physical inputs and 4 physical outputs:
3 AIFs/DAIs and one ADC/DAC pair. Internal routing is accomplished by
a 4-channel mixer connected to each output.
The analog and digital sides of the ADC/DAC are in separate ASoC
components, so card-level DAPM routes (provided in the device tree) are
necessary to connect them together. Currently, these routes are wrong.
For AIF1 Playback, the correct topology is:
||<<============ sun8i-codec ===========>>||
|| ||
CPU DAI -> AIF1 DA0 -> DAC Mixer -> DAC (digital) -> DAC (analog)
|| ||
but the driver and device trees currently describe:
|| ||
CPU DAI -> AIF1 DA0 -------------------------------> DAC (analog)
|| \--> DAC Mixer -> ??? [dead end] ||
For AIF1 Capture, there is an additional problem, because the Mixer
route is backward. The topology should be:
|| ||
ADC (analog) -> ADC (digital) -> AIF1 AD0 Mixer -> AIF1 AD0 -> CPU DAI
|| ||
but the driver and device trees currently describe:
|| ||
ADC (analog) -> AIF1 AD0 ------------------------------------> CPU DAI
|| \--> ADC Mixer -> ??? [dead end] ||
The ADC/DAC are only powered because AIF1 AD0 (capture) has supply
routes from the ADC, and AIF1 DA0 (playback) has supply routes from the
DAC. However, neither set of supply routes matches the hardware
topology. Audio can be routed among AIF1/2/3 without using the ADC or
DAC at all; and audio can be routed from the ADC to the DAC without
using any AIFs (via the "ADC Digital DAC Playback Switch"). Because the
DAPM routes are wrong, both of these use cases are currently broken.
This commit adds the necessary widgets and routes to represent the real
hardware topology, with functionality equivalent to the current driver.
For the existing "allwinner,sun8i-a33-codec" compatible, widgets with
the old names are kept as wrappers around the new widgets, so existing
device trees will continue to work. For "allwinner,sun50i-a64-codec",
the old widgets can be omitted, because no device trees yet use that
compatible.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200726012557.38282-3-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
As new function fsl_sai_dir_is_synced is included for checking if
stream is synced by the opposite stream, then replace the existing
synchronous checking with this new function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-4-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Tx synchronous with Rx: The RMR is the word mask register, it is used
to mask any word in the frame, it is not relating to clock generation,
So it is no need to be changed when Tx is going to be enabled.
Rx synchronous with Tx: The TMR is the word mask register, it is used
to mask any word in the frame, it is not relating to clock generation,
So it is no need to be changed when Rx is going to be enabled.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-3-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code enables TCSR.TE and RCSR.RE together, and disable
TCSR.TE and RCSR.RE together in trigger(), which only supports
one operation mode：
1. Rx synchronous with Tx: TE is last enabled and first disabled
Other operation mode need to be considered also:
2. Tx synchronous with Rx: RE is last enabled and first disabled.
3. Asynchronous mode: Tx and Rx are independent.
So the enable TCSR.TE and RCSR.RE sequence and the disable
sequence need to be refined accordingly for #2 and #3.
There is slightly against what RM recommennds with this change.
For example in Rx synchronous with Tx mode, case "aplay 1.wav;
arecord 2.wav" enable TE before RE. But it should be safe to
do so, judging by years of testing results.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20200805063413.4610-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_get_irq_byname() is used when there is list
of interrupts in the device node. As lpass-platform
has only one interrupt entry, use platform_get_irq()
instead.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Link: https://lore.kernel.org/r/1597402388-14112-12-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_get_resource_byname() is used when there
is list of reg entries. As lpass-cpu node has only
one reg entry, use platform_get_resource() instead.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-11-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
i2sctl register value is set to 0 during hw_free(). This
impacts any ongoing concurrent session on the same i2s
port. As trigger() stop already resets enable bit to 0,
there is no need of explicit hw_free. Removing it to
fix the issue.
Fixes: 80beab8e1d ("ASoC: qcom: Add LPASS CPU DAI driver")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-7-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
I2SCTL and DMACTL registers has different bits alignment for newer
LPASS variants of SC7180 soc. Use REG_FIELD_ID() to define the
reg_fields in platform specific file and removed shifts and mask
macros for such registers from header file.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Link: https://lore.kernel.org/r/1597402388-14112-6-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
lpass_pcm_data is never freed. Free it in close
ops to avoid memory leak.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-5-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We are allocating dma memory for component->dev but trying to mmap
such memory for substream->pcm->card->dev. Replace device argument
in mmap with component->dev to fix this.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-4-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Ahbix clock is optional clock and not needed for all platforms.
Move it to lpass-apq8016/ipq806x as it is not needed for sc7180.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-3-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
LPASS variants have their own soc specific clocks that needs to be
enabled for MI2S audio support. Added a common variable in drvdata to
initialize such clocks using bulk clk api. Such clock names is
defined in variants specific data and needs to fetched during init.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-2-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The (new?) style of clk registration uses clk_hw based APIs so that we
can more easily see the difference between clk providers and clk
consumers. Use the clk_hw based APIs to do this and migrate to devm for
the clkdev creation so that we can reduce the amount of code.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-4-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The __clk_get_name() API is deprecated. Use clk_hw_get_name() or
proper registration techniques to avoid it.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-3-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
I see a spew of "sysclk/dai not set correctly" whenever I cat
/sys/kernel/debug/clk/clk_summary on my device. This is because the
master pointer isn't set yet in this driver. A user isn't going to be
able to do much if this check is failing so this error message isn't
really an error, it's more of a kernel debug message. Lower the priority
to dev_dbg() so that it isn't so noisy.
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200804000531.920688-2-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When power_up_sst() fails, stream needs to be freed
just like when try_module_get() fails. However, current
code is returning directly and ends up leaking memory.
Fixes: 0121327c1a ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver supports WM1811, WM8994, WM8958 devices but according to
documentation and the regmap definitions the WM8958_DSP2_* registers
are only available on WM8958. In current code these registers are
being accessed as if they were available on all the three chips.
When starting playback on WM1811 CODEC multiple errors like:
"wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5"
can be seen, which is caused by attempts to read an unavailable
WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent
commit "e2329ee ASoC: soc-component: add soc_component_err()".
This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling
is only done for WM8958.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason interrupt set and clear register offsets are
not set correctly.
This patch corrects them!
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200811103452.20448-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC2 and DAC2 are not available on WM1811 device. This patch moves
the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing
them and avoid unreadable register access on WM1811.
This allows to get rid of warnings during boot like:
wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20200804141043.11425-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As the recent fix addressed the channel swap problem more properly,
update the comment as well.
Fixes: 1b7ecc241a ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109")
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.
Fixes: 11e424e88b ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB")
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable.vger.kernel.org>
Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All device-specific small fixes and quirks mostly for usual
suspects, USB-audio and HD-audio.
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Merge tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All device-specific small fixes and quirks mostly for usual suspects,
USB-audio and HD-audio"
* tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: echoaudio: Fix potential Oops in snd_echo_resume()
ALSA: hda/hdmi: Use force connectivity quirk on another HP desktop
ALSA: hda/realtek - Fix unused variable warning
ALSA: hda - reverse the setting value in the micmute_led_set
ALSA: echoaduio: Drop superfluous volatile modifier
ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control
ALSA: usb-audio: add quirk for Pioneer DDJ-RB
ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109
ALSA: hda - fix the micmute led status for Lenovo ThinkCentre AIO
ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109
ALSA: hda/realtek: Fix pin default on Intel NUC 8 Rugged
ALSA: usb-audio: Creative USB X-Fi Pro SB1095 volume knob support
ALSA: usb-audio: fix spelling mistake "buss" -> "bus"
Freeing chip on error may lead to an Oops at the next time
the system goes to resume. Fix this by removing all
snd_echo_free() calls on error.
Fixes: 47b5d028fd ("ALSA: Echoaudio - Add suspend support #2")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200813074632.17022-1-dinghao.liu@zju.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's another HP desktop has buggy BIOS which flags the Port
Connectivity bit as no connection.
Apply force connectivity quirk to enable DP/HDMI audio.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200811095336.32396-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix forgot to remove the unused variable that triggers a
compile warning now:
sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable]
Fix it.
Fixes: 404690649e ("ALSA: hda - reverse the setting value in the micmute_led_set")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of the DAPM widgets for DSP ASoC components reuse reg field
of the widgets for its internal calculations, however these are not
real registers. So read/writes to these numbers are not really
valid. However ASoC core will read these registers to get default
state during startup.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
To fix this add dummy read/write function to return default value.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like the q6afe-dai dapm widget registers are set as "0",
which is a not correct.
As this registers will be read by ASoC core during startup
which will throw up errors, Fix this by making the registers
as SND_SOC_NOPM as these should be never used.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Along with the recent unification of snd_soc_component_read*()
functions, the behavior of snd_soc_component_read() was changed
slightly; namely it returns the register read value directly, and even
if an error happens, it returns zero (but it prints an error
message). That said, the caller side can't know whether it's an error
or not any longer.
Ideally this shouldn't matter much, but in practice this seems causing
a regression, as John reported. And, grepping the tree revealed that
there are still plenty of callers that do check the error code, so
we'll need to deal with them in anyway.
As a quick band-aid over the regression, this patch changes the return
value of snd_soc_component_read() again to the negative error code.
It can't work, obviously, for 32bit register values, but it should be
enough for the known regressions, so far.
Fixes: cf6e26c71b ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200810134631.19742-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Before the micmute_led_set() is introduced, the function of
alc_gpio_micmute_update() will set the gpio value with the
!micmute_led.led_value, and the machines have the correct micmute led
status. After the micmute_led_set() is introduced, it sets the gpio
value with !!micmute_led.led_value, so the led status is not correct
anymore, we need to set micmute_led_polarity = 1 to workaround it.
Now we fix the micmute_led_set() and remove micmute_led_polarity = 1.
Fixes: 87dc36482c ("ALSA: hda/realtek - Add LED class support for micmute LED")
Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dsp_registers field of struct echoaduio has the volatile modifier,
but it's basically superfluous; the field is accessed only for the
base pointer of readl() and writel(), hence marking with __iomem alone
should suffice. OTOH, having the volatile prefix causes a compile
warning like:
sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers]
So it's better to drop this superfluous modifier.
Link: https://lore.kernel.org/r/20200803143958.24324-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replacing string compare with "codec_dai->name" instead of comparing with
"codec_dai->component->name" in hw_params because,
Here the component name for codec RT1015 is "i2c-10EC5682:00"
and will never be "rt1015-aif1" as it is codec-dai->name.
So, strcmp() always compares and fails to set the
sysclk,pll,bratio for expected codec-dai="rt1015-aif1".
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200807161046.17932-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.
Disable the volume control to workaround the issue.
Fixes: f8c11eb7da ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200810133108.31580-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200810021659.7429-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
* API cleanups and conversions to the unified mute_stream() call
* Simplify I/O helper functions
* Use helper macros to retrieve RTD from substreams
ASoC drivers:
* Lots of fixes and cleanups in Intel ASoC drivers
* Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
* Minor code refacotring for SG-buffer handling
HD-audio:
* Generalization of mute-LED handling with LED classdev
* Intel silent stream support for HDMI
* Device-specific fixes: CA0132, Loongson-3
Others:
* Usual USB- and HD-audio quirks for various devices
* Fixes for echoaudio DMA position handling
* Various documents and trivial fixes for sparse warnings
* Conversion to adapt inclusive terms
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Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
- API cleanups and conversions to the unified mute_stream() call
- Simplify I/O helper functions
- Use helper macros to retrieve RTD from substreams
ASoC drivers:
- Lots of fixes and cleanups in Intel ASoC drivers
- Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
- Minor code refacotring for SG-buffer handling
HD-audio:
- Generalization of mute-LED handling with LED classdev
- Intel silent stream support for HDMI
- Device-specific fixes: CA0132, Loongson-3
Others:
- Usual USB- and HD-audio quirks for various devices
- Fixes for echoaudio DMA position handling
- Various documents and trivial fixes for sparse warnings
- Conversion to adopt inclusive terms"
* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
ALSA: pci: delete repeated words in comments
ALSA: isa: delete repeated words in comments
ALSA: hda/tegra: Add 100us dma stop delay
ALSA: hda: Add dma stop delay variable
ASoC: hda/tegra: Set buffer alignment to 128 bytes
ALSA: seq: oss: Serialize ioctls
ALSA: hda/hdmi: Add quirk to force connectivity
ALSA: usb-audio: add startech usb audio dock name
ALSA: usb-audio: Add support for Lenovo ThinkStation P620
Revert "ALSA: hda: call runtime_allow() for all hda controllers"
ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
ALSA: docs: fix typo
ALSA: doc: use correct config variable name
ASoC: core: Two step component registration
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
...
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a spelling mistake in a usb_audio_dbg debug message. Also
replace "param" with "parameter". Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra HDA has audio data buffer for upto tens of frames, this buffer
can help to avoid underflow. HW will keep issuing new data fetch
request when buffers are not full and current BDL is not done. When SW
disable DMA RUN bit for a stream, HW can't cancel the already issued data
fetch request and hence it can't stop DMA. HW has to wait for all issued
data fetch request get data returned before it stops DMA.
This HW behavior is not in sync with HDA spec which says DMA RUN bit
should be cleared within 1 audio frame. For Tegra, DMA RUN bit was
active for more than one audio frame, due to this the timeout in
snd_hdac_stream_sync function is not helping. When Stream reset set
and clear happens during DMA RUN bit active state it results in Memory
Decode error.
Unfortunately, there is no way to detect when these data accesses have
completed, but testing has shown that a 100us delay between Stream reset
set and clear operation for Tegra avoids the memory decode error.
Therefore, adding a 100us dma stop delay.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-4-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A variable dma_stop_delay is added as a new member in hdac_bus
structure to avoid memory decode error incase DMA RUN bit is not
disabled in the given timeout from snd_hdac_stream_sync function and
followed by stream reset which results in memory decode error between
reset set and clear operation.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer
alignment to be multiple of 128 bytes. This fix is applied as gstreamer
alsasink gets stuck with the default buffer-time and latency-time
parameters with 4 byte buffer alignment.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases. This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.
Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency. There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.
Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com
Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com
Suggested-by: Hillf Danton <hdanton@sina.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Prepare for tasklet API modernization (Romain Perier, Allen Pais, Kees Cook)
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Merge tag 'tasklets-v5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux
Pull tasklets API update from Kees Cook:
"These are the infrastructure updates needed to support converting the
tasklet API to something more modern (and hopefully for removal
further down the road).
There is a 300-patch series waiting in the wings to get set out to
subsystem maintainers, but these changes need to be present in the
kernel first. Since this has some treewide changes, I carried this
series for -next instead of paining Thomas with it in -tip, but it's
got his Ack.
This is similar to the timer_struct modernization from a while back,
but not nearly as messy (I hope). :)
- Prepare for tasklet API modernization (Romain Perier, Allen Pais,
Kees Cook)"
* tag 'tasklets-v5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux:
tasklet: Introduce new initialization API
treewide: Replace DECLARE_TASKLET() with DECLARE_TASKLET_OLD()
usb: gadget: udc: Avoid tasklet passing a global
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name
and shows up currently as "USB Sound Device" in ALSA.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull crypto updates from Herbert Xu:
"API:
- Add support for allocating transforms on a specific NUMA Node
- Introduce the flag CRYPTO_ALG_ALLOCATES_MEMORY for storage users
Algorithms:
- Drop PMULL based ghash on arm64
- Fixes for building with clang on x86
- Add sha256 helper that does the digest in one go
- Add SP800-56A rev 3 validation checks to dh
Drivers:
- Permit users to specify NUMA node in hisilicon/zip
- Add support for i.MX6 in imx-rngc
- Add sa2ul crypto driver
- Add BA431 hwrng driver
- Add Ingenic JZ4780 and X1000 hwrng driver
- Spread IRQ affinity in inside-secure and marvell/cesa"
* 'linus' of git://git.kernel.org/pub/scm/linux/kernel/git/herbert/crypto-2.6: (157 commits)
crypto: sa2ul - Fix inconsistent IS_ERR and PTR_ERR
hwrng: core - remove redundant initialization of variable ret
crypto: x86/curve25519 - Remove unused carry variables
crypto: ingenic - Add hardware RNG for Ingenic JZ4780 and X1000
dt-bindings: RNG: Add Ingenic RNG bindings.
crypto: caam/qi2 - add module alias
crypto: caam - add more RNG hw error codes
crypto: caam/jr - remove incorrect reference to caam_jr_register()
crypto: caam - silence .setkey in case of bad key length
crypto: caam/qi2 - create ahash shared descriptors only once
crypto: caam/qi2 - fix error reporting for caam_hash_alloc
crypto: caam - remove deadcode on 32-bit platforms
crypto: ccp - use generic power management
crypto: xts - Replace memcpy() invocation with simple assignment
crypto: marvell/cesa - irq balance
crypto: inside-secure - irq balance
crypto: ecc - SP800-56A rev 3 local public key validation
crypto: dh - SP800-56A rev 3 local public key validation
crypto: dh - check validity of Z before export
lib/mpi: Add mpi_sub_ui()
...
The variable rtd was left unused in psc_dma_free(), even unnoticed
during conversion to a new style:
sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable]
Drop the superfluous one.
Fixes: 6d1048bc11 ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803144630.9615-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_i2s.c:167:12: warning: 'tegra210_i2s_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_i2s.c:179:12: warning: 'tegra210_i2s_runtime_resume' defined but not used [-Wunused-function]
Fixes: c0bfa98349 ("ASoC: tegra: Add Tegra210 based I2S driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_dmic.c:43:12: warning: 'tegra210_dmic_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_dmic.c:55:12: warning: 'tegra210_dmic_runtime_resume' defined but not used [-Wunused-function]
Fixes: 8c8ff982e9 ("ASoC: tegra: Add Tegra210 based DMIC driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:579:12: warning: 'tegra_ahub_runtime_resume' defined but not used [-Wunused-function]
Fixes: 16e1bcc2ca ("ASoC: tegra: Add Tegra210 based AHUB driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_admaif.c:232:12: warning: 'tegra_admaif_runtime_resume' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
Fixes: f74028e159 ("ASoC: tegra: Add Tegra210 based ADMAIF driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra186_dspk.c:74:12: warning: 'tegra186_dspk_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra186_dspk.c:86:12: warning: 'tegra186_dspk_runtime_resume' defined but not used [-Wunused-function]
Fixes: 327ef64702 ("ASoC: tegra: Add Tegra186 based DSPK driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.
USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0
Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.
USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)
So turn off the FU to avoid the error.
Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200803142612.17156-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we found an issue about the suspend and resume. If dmic is
recording the sound, and we run suspend and resume, after the resume,
the dmic can't work well anymore. we need to close the app and reopen
the app, then the dmic could record the sound again.
For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2
-r 48000 test.wav", then suspend and resume, after the system resume
back, we speak to the dmic. then stop the arecord, use aplay to play
the test.wav, we could hear the sound recorded after resume is weird,
it is not what we speak to the dmic.
I found two registers are set in the dai_hw_params(), if the two
registers are set during the resume, this issue could be fixed.
Move the code of the dai_hw_params() into the pdm_dai_trigger(), then
these two registers will be set during resume since pdm_dai_trigger()
will be called during resume. And delete the empty function
dai_hw_params().
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200730123138.5659-1-hui.wang@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause
audio distortion.
This change was sent as a comment below the --- line when submitting
commit 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so
it was not supposed to get merged.
Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to
fix the regression.
Fixes: 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE")
Reported-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200803115233.19034-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this case:
aplay -Dhw:x 16khz.wav 24khz.wav
There is sound distortion for 24khz.wav. The reason is that setting
PLL of WM8962 with set_bias_level function, the bias level is not
changed when 24khz.wav is played, then the PLL won't be reset, the
clock is not correct, so distortion happens.
The resolution of this issue is to remove fsl_asoc_card_set_bias_level.
Move PLL configuration to hw_params and hw_free.
After removing fsl_asoc_card_set_bias_level, also test WM8960 case,
it can work.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1596420811-16690-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
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Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.9
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
This reverts commit 9a6418487b ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Fixes: 9a6418487b ("ALSA: hda: call runtime_allow() for all hda controllers")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ca0113 command had the wrong group_id, 0x48 when it should've been
0x30. The front microphone selection should now work.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-1-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a mechanism for true two-step component registration. This
mimics device registration flow where initialization is the first step
while addition goes as second in line. Drivers may choose to modify
component's fields before registering component to ASoC subsystem via
snd_soc_add_component.
Patchset achieves status quo - behavior of snd_soc_register_component
remains unchanged.
Cezary Rojewski (3):
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Two step component registration
include/sound/soc-component.h | 3 --
include/sound/soc.h | 11 +++---
sound/soc/soc-component.c | 16 ---------
sound/soc/soc-core.c | 52 +++++++++++++++++----------
sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++---
sound/soc/stm/stm32_adfsdm.c | 9 +++--
6 files changed, 55 insertions(+), 50 deletions(-)
--
2.17.1
Modify snd_soc_add_component so it calls snd_soc_component_initialize
no longer and thus providing true two-step registration. Drivers may
choose to change component's fields before actually adding it to ASoC
subsystem.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move 'name' field initialization responsibility back to
snd_soc_component_initialize to prepare snd_soc_add_component function
for being called separatelly as a second registration step.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To allow for two-step component registration, expose
snd_soc_component_initialize function and move it back to soc-core.c.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enabling a whole subsystem from a single driver 'select' is frowned
upon and won't be accepted in new drivers, that need to use 'depends on'
instead. Existing selection of DMADEVICES will then cause circular
dependencies. Replace them with a dependency.
Signed-off-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200731152433.1297-3-laurent.pinchart@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The fifo_depth is 64 on i.MX8QM/i.MX8QXP, 128 on i.MX8MQ, 16 on
i.MX7ULP.
Original FSL_SAI_CR1_RFW_MASK value 0x1F is not suitable for
these platform, the FIFO watermark mask should be updated
according to the fifo_depth.
Fixes: a860fac420 ("ASoC: fsl_sai: Add support for imx7ulp/imx8mq")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1596176895-28724-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit b73287f0b0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
changed the meaning of dpcm_playback/dpcm_capture and now requires the
CPU DAI BE to aligned with those flags.
This broke all Amlogic cards with uni-directional backends (All gx and
most axg cards).
While I'm still confused as to how this change is an improvement, those
cards can't remain broken forever. Hopefully, next time an API change is
done like that, all the users will be updated as part of the change, and
not left to fend for themselves.
Fixes: b73287f0b0 ("ASoC: soc-pcm: dpcm: fix playback/capture checks")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200731120603.2243261-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Previous updates to set dailink capabilities and check dailink
capabilities were based on a flawed assumption that all dais support
the same capabilities as the dailink. This is true for TDM
configurations but existing configurations use an amplifier and a
capture device on the same dailink, and the tests would prevent the
card from probing.
This patch modifies the snd_soc_dai_link_set_capabilities()
helper so that the dpcm_playback (resp. dpcm_capture) dailink
capabilities are set if at least one dai supports playback (resp. capture).
Likewise the checks are modified so that an error is reported only
when dpcm_playback (resp. dpcm_capture) is set but none of the CPU
DAIs support playback (resp. capture).
Fixes: 25612477d2 ('ASoC: soc-dai: set dai_link dpcm_ flags with a helper')
Fixes: b73287f0b0 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Suggested-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200723180533.220312-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few wrap-up small fixes for the usual HD-audio and USB-audio stuff:
- A regression fix for S3 suspend on old Intel platforms
- A fix for possible Oops in ASoC HD-audio binding
- Trivial quirks for various devices
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Merge tag 'sound-5.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few wrap-up small fixes for the usual HD-audio and USB-audio stuff:
- A regression fix for S3 suspend on old Intel platforms
- A fix for possible Oops in ASoC HD-audio binding
- Trivial quirks for various devices"
* tag 'sound-5.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fixed HP right speaker no sound
ALSA: hda: fix NULL pointer dereference during suspend
ALSA: hda/hdmi: Fix keep_power assignment for non-component devices
ALSA: hda: Workaround for spurious wakeups on some Intel platforms
ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)
ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
ALSA: usb-audio: Add implicit feedback quirk for SSL2
The various list iterators are able to handle an empty list.
The only effect of avoiding the loop is not initializing some
index variables.
Drop list_empty tests in cases where these variables are not
used.
The semantic patch that makes these changes is as follows:
(http://coccinelle.lip6.fr/)
<smpl>
@@
expression x,e;
iterator name list_for_each_entry;
statement S;
identifier i;
@@
-if (!(list_empty(x))) {
list_for_each_entry(i,x,...) S
- }
... when != i
? i = e
@@
expression x,e;
iterator name list_for_each_entry_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_entry_safe(i,j,x,...) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
@@
expression x,e;
iterator name list_for_each;
statement S;
identifier i;
@@
-if (!(list_empty(x))) {
list_for_each(i,x) S
- }
... when != i
? i = e
@@
expression x,e;
iterator name list_for_each_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_safe(i,j,x) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
// -------------------
@@
expression x,e;
statement S;
identifier i;
@@
-if (!(list_empty(x)))
list_for_each_entry(i,x,...) S
... when != i
? i = e
@@
expression x,e;
statement S;
identifier i,j;
@@
-if (!(list_empty(x)))
list_for_each_entry_safe(i,j,x,...) S
... when != i
when != j
(
i = e;
|
? j = e;
)
@@
expression x,e;
statement S;
identifier i;
@@
-if (!(list_empty(x)))
list_for_each(i,x) S
... when != i
? i = e
@@
expression x,e;
statement S;
identifier i,j;
@@
-if (!(list_empty(x)))
list_for_each_safe(i,j,x) S
... when != i
when != j
(
i = e;
|
? j = e;
)
</smpl>
---
drivers/media/pci/saa7134/saa7134-core.c | 14 ++---
drivers/media/usb/cx231xx/cx231xx-core.c | 16 ++----
drivers/media/usb/tm6000/tm6000-core.c | 24 +++-------
drivers/net/ethernet/mellanox/mlx5/core/steering/dr_matcher.c | 13 ++---
drivers/net/ethernet/mellanox/mlx5/core/steering/dr_rule.c | 5 --
drivers/net/ethernet/sfc/ptp.c | 20 +++-----
drivers/net/wireless/ath/dfs_pattern_detector.c | 15 ++----
sound/soc/intel/atom/sst/sst_loader.c | 10 +---
sound/soc/intel/skylake/skl-pcm.c | 8 +--
sound/soc/intel/skylake/skl-topology.c | 5 --
10 files changed, 53 insertions(+), 77 deletions(-)
PulseAudio (and perhaps other userspace utilities) can not detect any
jack for rk3399_gru_sound as the driver doesn't expose related Jack
kcontrols.
This patch adds two DAPM pins to the headset jack, where the
snd_soc_card_jack_new() call automatically creates "Headphones Jack" and
"Headset Mic Jack" kcontrols from them.
With an appropriate ALSA UCM config specifying JackControl fields for
the "Headphones" and "Headset" (mic) devices, PulseAudio can detect
plug/unplug events for both of them after this patch.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721182709.6895-1-alpernebiyasak@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the WM8962 datasheet, there is no register at address 0x200.
WM8962_GPIO_BASE is just a base address for the GPIO registers and not a
real register, so remove it from wm8962_readable_register().
Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip
its access.
This fixes the following errors:
wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16
wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200717135959.19212-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use resource_size rather than a verbose computation on
the end and start fields.
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
<smpl>
@@ struct resource ptr; @@
- (ptr.end - ptr.start + 1)
+ resource_size(&ptr)
</smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Link: https://lore.kernel.org/r/1595751933-4952-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
list_for_each_entry_safe is able to handle an empty list.
The only effect of avoiding the loop is not initializing the
index variable.
Drop list_empty tests in cases where these variables are not
used.
Note that list_for_each_entry_safe is defined in terms of
list_first_entry, which indicates that it should not be used on an
empty list. But in list_for_each_entry_safe, the element obtained by
list_first_entry is not really accessed, only the address of its
list_head field is compared to the address of the list head, so the
list_first_entry is safe.
The semantic patch that makes this change is as follows (with another
variant for the no brace case): (http://coccinelle.lip6.fr/)
<smpl>
@@
expression x,e;
iterator name list_for_each_entry_safe;
statement S;
identifier i,j;
@@
-if (!(list_empty(x))) {
list_for_each_entry_safe(i,j,x,...) S
- }
... when != i
when != j
(
i = e;
|
? j = e;
)
</smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Link: https://lore.kernel.org/r/1595761112-11003-2-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Passing specific snd_soc_card structure depending on the ACPI ID.
In future we can add other IDs in the ACPI table and pass the structure.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200728160255.31020-3-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As in future our machine driver supports multiple codecs
So changing naming convention of snd_soc_card struct and its fields.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200728160255.31020-2-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for voice and BT calls, along with standard
audio output via the speaker, earpiece, headphone jack, HDMI, and
any accessories compatible with Midas boards. This patch also supports
headphone/headset detection and headsets with inline buttons.
[m.szyprowski: adaptation to v5.1+ kernels (DAI links initialization)]
[s.nawrocki: removal of the clk API calls for CODEC MCLK, the jack data
structure moved to struct midas_priv, coding style and typo fixes,
conversion to new cpu/codec/dai-node binding]
Signed-off-by: Simon Shields <simon@lineageos.org>
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200728131111.14334-2-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reset the device before programming the registers or all programming
will be lost as the device resets registers to default settings.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200730142419.28205-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The header was updated to align with the data sheet to start the GPO_CFG
at GPO_CFG0. The code was not updated to the change and therefore the
GPO_CFG0 register was not written to.
Fixes: 6617cff6a0 ("ASoC: tlv320adcx140: Add GPO configuration and drive output config")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200730142419.28205-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All channels are enabled at boot up, this patch ensures that all
channels are disabled at boot and whenever the function is called.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200730055319.1522-3-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable 8kHz audio support for Intel Keem Bay platform.
Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit@intel.com>
Reviewed-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200730055319.1522-2-michael.wei.hong.sit@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The allocation order of things in soc_new_pcm_runtime was changed to
move the device_register before the allocation of the rtd structure.
This was to allow the rtd allocation to be managed by devm. However
currently the sysfs entries are added by device_register and their
visibility depends on variables within the rtd structure, this causes
the pmdown_time and dapm_widgets sysfs entries to be missing for all
rtds.
Correct this issue by manually calling device_add_groups after the
appropriate information is available.
Fixes: d918a37610 ("ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200730120715.637-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Standard dai format property don't need the "amlogic," prefix.
There nothing amlogic specific about them. Just remove it.
Fixes: 435857e015 ("ASoC: meson: align axg card driver with DT bindings documentation")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After carefully checking, it appears that both tdmout and tdmin require the
rising edge of the sclk they get to be synchronized with the frame sync
event (which should be a rising edge of lrclk).
TDMIN was improperly set before this patch. Remove the sclk_invert quirk
which is no longer needed and fix the sclk phase.
Fixes: 1a11d88f49 ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
After carefully checking the result provided by the TDMIN on the g12a and
sm1 SoC families, the TDMIN skew offset appears to be 3 instead of 2 on the
axg.
Fixes: f01bc67f58 ("ASoC: meson: axg-tdm-formatter: rework quirks settings")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The .set_fmt() callback of the axg tdm interface incorrectly
test the content of SND_SOC_DAIFMT_MASTER_MASK as if it was a
bitfield, which it is not.
Implement the test correctly.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This converts all the existing DECLARE_TASKLET() (and ...DISABLED)
macros with DECLARE_TASKLET_OLD() in preparation for refactoring the
tasklet callback type. All existing DECLARE_TASKLET() users had a "0"
data argument, it has been removed here as well.
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Kees Cook <keescook@chromium.org>
HP NB right speaker had no sound output.
This platform was connected to I2S Amp for speaker out.(None Realtek I2S Amp IC)
EC need to check codec GPIO1 pin to initial I2S Amp.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/01285f623ac7447187482fb4a8ecaa7c@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add General Purpose Output (GPO) configuration and driver output
configuration. The GPOs can be configured as a GPO, IRQ, SDOUT or a
PDMCLK output. In addition the output drive can be configured with
various configurations.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200728160833.24130-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix white space issues and remove else case where it was not needed.
Convert "static const char *" to "static const char * const"
Fixes: 689c7655b5 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200728164339.16841-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the ASoC card registration fails and the codec component driver
never probes, the codec device is not initialized and therefore
memory for codec->wcaps is not allocated. This results in a NULL pointer
dereference when the codec driver suspend callback is invoked during
system suspend. Fix this by returning without performing any actions
during codec suspend/resume if the card was not registered successfully.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200728231011.1454066-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This set of patches is required for facilitating system S0ix
entry when the DSP is in D0I3. This first patch adds the missing
CORB/RIRB DMA stop and restart to the suspend/resume sequence along
with powering up/down the links. The second patch ensures that the
FW traces are disabled when the system enters S0ix with the DSP in D0I3.
Marcin Rajwa (2):
ASoC: SOF: Intel: fix the suspend procedure to support s0ix entry
ASoC: SOF: Intel: disable traces when switching to S0Ix D0I3
sound/soc/sof/intel/hda-dsp.c | 48 ++++++++++++++++++++++++++++++++---
1 file changed, 44 insertions(+), 4 deletions(-)
--
2.25.1
Update the shutdown GPIO property to be shutdown from shut-down.
Fixes: c173dba44c ("ASoC: tas2562: Introduce the TAS2562 amplifier")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200723160838.9738-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should always disable DMA trace on S0Ix. When staying at S0-D0I3,
we should enable DMA trace while both DMA Trace debug is enabled and
hda_enable_trace_D0I3_S0 is set. This commit corrects the existed
logic errors about that.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200727183613.1419005-3-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes the suspend & resume procedure to allow entry into the
low power states with some streams being active as a wake source - wake on
voice is a perfect example. The current implementation does not stop
the CORB/RIRB DMA and does not power down the HDA links. With firmware's
help, the platform has been able to still enter s0ix state on older
platforms, but the sequence is still incorrect, and the additional
driver actions are needed to ensure correct s0ix behaviour.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200727183613.1419005-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It's been reported that, when neither nouveau nor Nvidia graphics
driver is used, the screen starts flickering. And, after comparing
between the working case (stable 4.4.x) and the broken case, it turned
out that the problem comes from the audio component binding. The
Nvidia and AMD audio binding code clears the bus->keep_power flag
whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that
the component is actually bound, but it merely indicates that it's
ready for binding. So, when both nouveau and Nvidia are blacklisted
or not ready, the driver keeps running without the audio component but
also with bus->keep_power = false. This made the driver runtime PM
kicked in and powering down when unused, which results in flickering
in the graphics side, as it seems.
For fixing the bug, this patch moves the bus->keep_power flag change
into generic_acomp_notifier_set() that is the function called from the
master_bind callback of component ops; i.e. it's guaranteed that the
binding succeeded.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609
Fixes: 5a858e79c9 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200728082033.23933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've received a regression report on Intel HD-audio controller that
wakes up immediately after S3 suspend. The bisection leads to the
commit c4c8dd6ef8 ("ALSA: hda: Skip controller resume if not
needed"). This commit replaces the system-suspend to use
pm_runtime_force_suspend() instead of the direct call of
__azx_runtime_suspend(). However, by some really mysterious reason,
pm_runtime_force_suspend() causes a spurious wakeup (although it calls
the same __azx_runtime_suspend() internally).
As an ugly workaround for now, revert the behavior to call
__azx_runtime_suspend() and __azx_runtime_resume() for those old Intel
platforms that may exhibit such a problem, while keeping the new
standard pm_runtime_force_suspend() and pm_runtime_force_resume()
pair for the remaining chips.
Fixes: c4c8dd6ef8 ("ALSA: hda: Skip controller resume if not needed")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Right now the direction of a DAI has to be specified as a literal
number in the device tree, e.g.:
dai@0 {
reg = <0>;
direction = <2>;
};
but this does not make it immediately clear that this is a
playback/RX-only DAI.
Actually, q6asm-dai.c has useful defines for this. Move them to the
dt-bindings header to allow using them in the dts(i) files.
The example above then becomes:
dai@0 {
reg = <0>;
direction = <Q6ASM_DAI_RX>;
};
which is immediately recognizable as playback/RX-only DAI.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200727082502.2341-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
PME_EN state needs to restored to the value set by fmw.
For the devices which are not using I2S wake event which gets
enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount
of power as it blocks low power state.
For the devices using I2S wake event, PME_EN gets enabled in fmw and the
state should be maintained after ACP Power On.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Link: https://lore.kernel.org/r/20200724195600.11798-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87tuxtydcz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v9i9yddc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
runtime_usage of sound card has been observed to grow without bound.
For example:
$ cat /sys/devices/platform/sound/power/runtime_usage
46
$ sox -n -t s16 -r 48000 -c 2 - synth 1 sine 440 vol 0.1 | \
aplay -q -D hw:0,0 -f S16_LE -r 48000 -c 2
$ cat /sys/devices/platform/sound/power/runtime_usage
52
Commit 4e872a4682 ("ASoC: dapm: Don't force card bias level to be
updated") stops to force update bias_level on card. If card doesn't
provide set_bias_level callback, the snd_soc_dapm_set_bias_level()
is equivalent to NOP for card device.
As a result, dapm_pre_sequence_async() doesn't change the bias_level of
card device correctly. Thus, pm_runtime_get_sync() would be called in
dapm_pre_sequence_async() without symmetric pm_runtime_put() in
dapm_post_sequence_async().
Don't call pm_runtime_* on card device.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200724070731.451377-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.
Fixes: ff53664daf ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289")
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140616.298892-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the moment we have two separate functions to parse the sound card
properties from the device tree: qcom_snd_parse_of() for DPCM and
apq8016_sbc_parse_of() without DPCM. These functions are almost identical
except for a few minor differences.
This patch set extends qcom_snd_parse_of() to handle links without DPCM,
so that we can use one common function for all (qcom) machine drivers.
Stephan Gerhold (7):
ASoC: qcom: Use devm for resource management
ASoC: qcom: common: Use snd_soc_dai_link_set_capabilities()
ASoC: q6afe: Remove unused q6afe_is_rx_port() function
ASoC: qcom: common: Support parsing links without DPCM
ASoC: qcom: common: Parse properties with "qcom," prefix
ASoC: qcom: apq8016_sbc: Use qcom_snd_parse_of()
ASoC: qcom: common: Avoid printing errors for -EPROBE_DEFER
sound/soc/qcom/Kconfig | 1 +
sound/soc/qcom/apq8016_sbc.c | 120 ++++-------------------------------
sound/soc/qcom/apq8096.c | 28 +-------
sound/soc/qcom/common.c | 58 ++++++++++-------
sound/soc/qcom/qdsp6/q6afe.c | 8 ---
sound/soc/qcom/qdsp6/q6afe.h | 1 -
sound/soc/qcom/sdm845.c | 40 ++----------
7 files changed, 59 insertions(+), 197 deletions(-)
--
2.27.0
Modify dsm_init sequence and dsm param bin check condition.
- Move dsm_init() to after amp init setting to
make sure dsm init is last setting.
- dsm param bin check condition changed for extended register setting.
Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200724060149.19261-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With commit e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
every error different for ENOTSUPP or EPROBE_DEFER will log an error.
However, as explained in snd_soc_get_dai_name(), this callback may error
to indicate that the DAI is not matched by the component tested. If the
device provides other components, those may still match. Logging an error
in this case is misleading.
Don't use soc_component_ret() in snd_soc_component_of_xlate_dai_name()
to avoid spamming the log.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200723142020.1338740-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
qcom_snd_parse_of() tends to produce lots of error messages during bootup:
MultiMedia1: error getting cpu dai name
This happens because the DAIs are not probed until the ADSP remoteproc
has booted, which takes a while. Until it is ready, snd_soc_of_get_dai_name()
returns -EDEFER_PROBE to retry probing later. This is perfectly normal,
so cleanup the kernel log a bit by not printing in case of -EPROBE_DEFER.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-8-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have updated qcom_snd_parse_of() to handle the device
tree bindings used for apq8016_sbc, update the apq8016_sbc driver
to use the common function and remove the duplicated code.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-7-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
The apq8016_sbc device tree binding uses a "qcom," vendor prefix
for all device tree properties, while qcom_snd_parse_of() uses the
same properties without a prefix.
In the future it would be nice to make this consistent, however,
for backwards compatibility we need to parse both names to allow
apq8016_sbc to use the common qcom_snd_parse_of() function.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-6-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
So far qcom_snd_parse_of() was only used to parse the device tree
for boards using the QDSP6 driver together with DPCM. apq8016_sbc
uses an almost identical version (apq8016_sbc_parse_of()) which
parses links without DPCM.
Given the similarity of the two functions it is useful to combine
these two. To allow using qcom_snd_parse_of() in apq8016_sbc we
need to support parsing links without DPCM as well.
This is pretty simple: A DPCM link in the device tree is defined using:
- DPCM frontend: "cpu"
- DPCM backend: "cpu", "platform" and "codec"
... while a link without DPCM has "cpu" and "codec" (but no "platform").
Add a few more if conditions to handle links without DPCM correctly.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-5-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 4a95737440 ("ASoc: q6afe: add support to get
port direction"), since the function is not needed anymore.
q6afe-dai already exposes the possible directions for a DAI through
the DAI capabilities (playback/capture-only DAI). Now we use
snd_soc_dai_link_set_capabilities() to infer the information
directly from the DAI capabilities.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-4-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a212008925 ("ASoC: qcom: common: set correct directions for dailinks")
introduced a call to q6afe_is_rx_port() to set the dpcm_playback/capture
parameters correctly. This is necessary because those parameters are now
validated to match the capabilities of the DAIs. [1]
The disadvantage of introducing the call to q6afe_is_rx_port() is that
it makes the qcom_snd_parse_of() helper dependent on the QDSP6 driver.
When the ADSP is bypassed (e.g. in apq8016-sbc) QDSP6 is not used.
There is a generic solution for this now: The correct direction for the links
is already defined by the DAI capabilities (e.g. rx ports only support playback).
Commit 25612477d2 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper")
introduced the snd_soc_dai_link_set_capabilities() function that we can use
to set dpcm_playback/dpcm_capture according to the capabilities of the DAIs.
Use that for both FE/BE DAI links to avoid the dependency on the QDSP6 driver.
[1]: https://lore.kernel.org/alsa-devel/20200616085409.GA110999@gerhold.net/
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify the machine drivers for newer SoCs a bit by using the
devm_* function calls that automatically release the resources
when the driver is removed or when probing fails.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200723183904.321040-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Global EN register guide to off before AMP_EN register
when amp disable sequence.
- remove AMP_EN control before max98390_dac_event call
Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200724060058.19201-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Support same propeties as simple card for configuring fmt
from DT.
In order to make this change compatible with old DT, these
properties are optional.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1595302910-19688-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ESAI interfaces may share same interrupt line with EDMA on
some platforms (e.g. i.MX8QXP, i.MX8QM).
Add IRQF_SHARED flag to allow sharing the irq among several
devices
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1595476808-28927-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Build errors are seen on 32-bit platforms because of a plain 64-by-32
division. For example, following build erros were reported.
"ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko]
undefined!"
"ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko]
undefined!"
This can be fixed by using div_u64() helper from 'math64.h' header.
Fixes: 8c8ff982e9 ("ASoC: tegra: Add Tegra210 based DMIC driver")
Reported-by: Geert Uytterhoeven <geert@linux-m68k.org>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595492011-2411-1-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_J721E_EVM should not select SND_SOC_PCM3168A_I2C when I2C
is not enabled. That causes build errors, so make this driver's
symbol depend on I2C.
WARNING: unmet direct dependencies detected for SND_SOC_PCM3168A_I2C
Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n]
Selected by [m]:
- SND_SOC_J721E_EVM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (DMA_OMAP [=y] || TI_EDMA [=m] || TI_K3_UDMA [=n] || COMPILE_TEST [=y]) && (ARCH_K3_J721E_SOC [=n] || COMPILE_TEST [=y])
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: data definition has no type or storage class
module_i2c_driver(pcm3168a_i2c_driver);
^~~~~~~~~~~~~~~~~
../sound/soc/codecs/pcm3168a-i2c.c:59:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int]
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: parameter names (without types) in function declaration
../sound/soc/codecs/pcm3168a-i2c.c:49:26: warning: ‘pcm3168a_i2c_driver’ defined but not used [-Wunused-variable]
static struct i2c_driver pcm3168a_i2c_driver = {
^~~~~~~~~~~~~~~~~~~
cc1: some warnings being treated as errors
Fixes: 6748d05590 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/e74c690c-c7f8-fd42-e461-4f33571df4ef@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718112403.13709-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Coefficient files now support additional metadata blocks, these
contain machine parsable text strings describing the parameters
contained in the coefficient file.
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200723110321.16382-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718111209.11760-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718110857.11520-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can use asoc_substream_to_rtd() macro,
let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8ob0yun.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc-xxx are getting rtd from substream by
rtd = substream->private_data;
But, getting data from "private_data" is very unclear.
This patch adds asoc_substream_to_rtd() macro which is
easy to understand that rtd from substream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wo2z0yve.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The series re-uses mt8183-mt6358-ts3a227-max98357.c to support machine driver
with max98357b.
The 1st patch enables left justified format from mt8183 audio platform.
The 2nd patch adds document for the new proposed compatible string for
max98357b.
The 3rd patch supports machine driver with max98357b and uses left justified
format for it.
Tzung-Bi Shih (3):
ASoC: mediatek: mt8183: support left justified format for I2S
ASoC: dt-bindings: mt8183: add compatible string for using max98357b
ASoC: mediatek: mt8183: support machine driver with max98357b
.../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 +
sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 ++++++++++++++++---
.../mt8183/mt8183-mt6358-ts3a227-max98357.c | 22 ++++++-
3 files changed, 73 insertions(+), 9 deletions(-)
--
2.28.0.rc0.105.gf9edc3c819-goog
Daniel Baluta <daniel.baluta@nxp.com>:
From: Daniel Baluta <daniel.baluta@nxp.com>
This patchseries contains a couple of SOF IMX fixes
found during our first IMX SOF release.
Daniel Baluta (7):
ASoC: SOF: define INFO_ flags in dsp_ops for imx8
ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value
ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X
ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M
ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X
ASoC: SOF: topology: Update SAI config bclk/fsync rate
ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs
sound/soc/sof/imx/imx8.c | 24 +++++++++++++++++++++---
sound/soc/sof/imx/imx8m.c | 4 ++--
sound/soc/sof/pcm.c | 8 ++++++++
sound/soc/sof/topology.c | 2 ++
4 files changed, 33 insertions(+), 5 deletions(-)
--
2.17.1
Commit 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI
startup/shutdown sequence"), introduced a slight change of semantics
to DAI startup/shutdown. If startup() returns an error, shutdown()
is now called for the DAI.
This causes a deadlock in hdac_hda which issues a call to
snd_hda_codec_pcm_put() in case open fails. Upon error, soc_pcm_open()
will call shutdown(), and pcm_put() ends up getting called twice. Result
is a deadlock on pcm->open_mutex, as snd_device_free() gets called from
within snd_pcm_open(). Typical task backtrace looks like this:
[ 334.244627] snd_pcm_dev_disconnect+0x49/0x340 [snd_pcm]
[ 334.244634] __snd_device_disconnect.part.0+0x2c/0x50 [snd]
[ 334.244640] __snd_device_free+0x7f/0xc0 [snd]
[ 334.244650] snd_hda_codec_pcm_put+0x87/0x120 [snd_hda_codec]
[ 334.244660] soc_pcm_open+0x6a0/0xbe0 [snd_soc_core]
[ 334.244676] ? dpcm_add_paths.isra.0+0x491/0x590 [snd_soc_core]
[ 334.244679] ? kfree+0x9a/0x230
[ 334.244686] dpcm_be_dai_startup+0x255/0x300 [snd_soc_core]
[ 334.244695] dpcm_fe_dai_open+0x20e/0xf30 [snd_soc_core]
[ 334.244701] ? snd_pcm_hw_rule_muldivk+0x110/0x110 [snd_pcm]
[ 334.244709] ? dpcm_be_dai_startup+0x300/0x300 [snd_soc_core]
[ 334.244714] ? snd_pcm_attach_substream+0x3c4/0x540 [snd_pcm]
[ 334.244719] snd_pcm_open_substream+0x69a/0xb60 [snd_pcm]
[ 334.244729] ? snd_pcm_release_substream+0x30/0x30 [snd_pcm]
[ 334.244732] ? __mutex_lock_slowpath+0x10/0x10
[ 334.244736] snd_pcm_open+0x1b3/0x3c0 [snd_pcm]
Fixes: 5bd70440cb ("ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2159
Link: https://lore.kernel.org/r/20200717101950.3885187-3-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdac_hda remove implementation fails to free the hda codec
resources, leading to memleaks at module unload. This gap has been there
from the start, commit 6bae5ea949 ("ASoC: hdac_hda: add asoc
extension for legacy HDA codec drivers").
Instead of duplicating the cleanup logic, use the common
snd_hda_codec_cleanup_for_unbind() to free the resources. Remove
existing code in hdac_hda to cleanup "codec.jackpoll_work" and call to
snd_hdac_regmap_exit(), as these are already done in
snd_hda_codec_cleanup_for_unbind().
The cleanup is done in ASoC component remove() callback and not in the
HDAC bus hdev_detach(). This is done to ensure the codec specific
cleanup routines are run before the parent card is freed.
Fixes: 6bae5ea949 ("ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2195
Link: https://lore.kernel.org/r/20200717101950.3885187-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error handling for patch_ops in hdac_hda_codec_probe().
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200717101950.3885187-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200719153822.59788-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Supports machine driver with max98357b
("mt8183-mt6358-ts3a227-max98357b").
The key difference from max98357a: max98357b needs to use left
justified format.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
MT8183 audio platform supports EIAJ and I2S formats. The code fixed to
use I2S format in the past.
Supports EIAJ mode via set_fmt ops and preserves to use I2S format as
the default format intentionally.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200720012559.906088-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Starting in commit cbc7a6b5a8 ("ASoC: soc-card: add
snd_soc_card_add_dai_link()"), error value from ASoc add_dai_link() is
no longer ignored.
The generic HDA machine driver relied on the old semantics to disable
i915 HDMI/DP audio codec at runtime. If no display codec was present,
add_dai_link() returned an error, but this was ignored and rest of the
card was successfully probed.
Fix the problem by changing the machine driver add_dai_link() to not
return an error in this case.
Fixes: cbc7a6b5a8 ("ASoC: soc-card: add snd_soc_card_add_dai_link()")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2261
Link: https://lore.kernel.org/r/20200714132804.3638221-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixup BE DAI links rate/channels parameters to match any values
from topology.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-8-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These parameters are read from topology file and sent to DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-7-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With SOF we support 1 ESAI interface and 1 SAI interface.
This patch adds SAI1 interface support existing on i.MX8/i.MX8X
boards.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-6-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, sai-port
is too generic. Physical DAI port on i.MX8MP is labeled SAI3.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-5-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This must match DAI name from topology. Also, esai-port is too generic
as they are 2 ESAIs on i.MX8/i.MX8X boards.
SOF integration only uses ESAI0 for now.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-4-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this change we no longer need to update num_drv when adding
new DAI driver.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-3-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the past, the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP were
defined in the SOF PCM core, but that was changed since
commit 27e322fabd ("ASoC: SOF: define INFO_ flags in dsp_ops")
Now these flags must be set in DSP ops.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200720072046.8152-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.
Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel that
sends/receives data to/from AHUB must intreface through an ADMAIF channel.
ADMA channel sending data to AHUB pairs with an ADMAIF Tx channel and
similarly ADMA channel receiving data from AHUB pairs with an ADMAIF Rx
channel. Buffer size is configurable for each ADMAIF channel, but currently
SW uses default values.
This patch registers ADMAIF driver with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes ADMAIF interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The ADMAIF device can be enabled in the DT via
"nvidia,tegra210-admaif" compatible binding.
Tegra PCM driver is updated to expose required PCM interfaces and
snd_pcm_ops callbacks.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-8-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the reset property name when allocating the GPIO descriptor.
The gpiod_get_optional appends either the -gpio or -gpios suffix to the
name.
Fixes: 1a476abc72 ("tas2770: add tas2770 smart PA kernel driver")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200720181202.31000-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Partially reverts commit 128f825aea ("ASoC: max98357a: move control
of SD_MODE to DAPM").
In order to have mute control of max98357 from machine drivers, commit
128f825aea ("ASoC: max98357a: move control of SD_MODE to DAPM")
moves the control of SD_MODE from DAI ops to DAPM events. However, pop
noise has been observed on rk3399-gru-kevin boards due to this commit.
The commit 128f825aea caused sequence of DAI clocks and SD_MODE
changed on rk3399-gru-kevin boards.
With the commit 128f825aeab7:
- SD_MODE will be set to 1 before DAI clocks start.
- SD_MODE will be set to 0 after DAI clocks stop.
As a result, pop noise.
Moves the control of SD_MODE back to DAI ops. In the meantime, uses an
additional flag in DAPM event to provide chance of mute control for
machine drivers.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Tested-By: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Link: https://lore.kernel.org/r/20200721114232.2812254-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This PR became fairly large, containing mostly the collection of
ASoC fixes that slipped from the previous request, so I sent now
a bit earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests
and fuzzing. The rest are other ASoC device-specific fixes (imx,
qcom, wm8974, amd, rockchip) as well as a trivial fix for a kernel
WARNING hit by syzkaller.
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Merge tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"This became fairly large, containing mostly the collection of ASoC
fixes that slipped from the previous request, so I sent now a bit
earlier than usual. But all changes look small and mostly
device-specific, hence nothing to worry too much.
Majority of changes are for x86 based platforms and their CODEC
drivers, in order to address some issues hit by their recent tests and
fuzzing. The rest are other ASoC device-specific fixes (imx, qcom,
wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING
hit by syzkaller"
* tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
ASoC: rt5682: Report the button event in the headset type only
ASoC: Intel: bytcht_es8316: Add missed put_device()
ASoC: rt5682: Enable Vref2 under using PLL2
ASoC: rt286: fix unexpected interrupt happens
ASoC: wm8974: remove unsupported clock mode
ASoC: wm8974: fix Boost Mixer Aux Switch
ASoC: SOF: core: fix null-ptr-deref bug during device removal
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: codecs: max98373: Removed superfluous volume control from chip default
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: fix kernel oops on route addition error
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
ASoC: Intel: bdw-rt5677: fix non BE conversion
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
ASoC: core: Remove only the registered component in devm functions
MAINTAINERS: Change Maintainer for some at91 drivers
ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
...
In commit d696a61413 ("ASoC: rt1015: Add condition to prevent SoC
providing bclk in ratio of 50 times of sample rate."), PLL input at 50fs
is no longer supported, the new recommended settings at 48Khz rate are:
PLL input SSP bclk
------------------------
64fs 3.073Mhz
100fs 4.8Mhz
(bclk update is reflected in topoplogy.)
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The mc_private->hdmi_pcm_list is populated by elements loaded during
DSP topology load. Valid topologies for this machine driver will always
have PCM nodes for HDMI, but driver should fail gracefully even in the case
this is not true. Add a sanity check to sof_sdw_hdmi_card_late_probe()
for this case. Without the fix, a null pcm handle gets dereferenced.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend the generic SOF Soundwire machine driver to support systems where
iDisp HDMI/DP audio codec is disabled for some reason (i915 driver
disabled, HDMI/DP implemented with a discrete GPU, etc). Switch codecs
to SoC dummy in the affected DAI links. This allows to reuse existing
topologies for this case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt711 jack detection properties are set from the machine drivers
during the card probe, as done in other ASoC examples.
KASAN reports a use-after-free error when unbinding drivers due to a
confusing sequence between the ACPI core, the device core and the
SoundWire device cleanups.
Rather than fixing this sequence, follow the recommendation to have
the same caller add and remove properties, add an explicit
device_remove_properties() in the card .remove() callback.
In future patches the use of device_add/remove_properties will be
replaced by a direct handling of a swnode, but the sequence will
remain the same.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We can get codec name from dai link.
Suggested-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Overview
========
Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio
Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio
pre-processing and post-processing. It also includes a programmable full
crossbar for routing audio data across these accelerators.
This series exposes some of these below mentioned HW devices as ASoC
components for Tegra platforms from Tegra210 onwards.
* ADMAIF : The interface between ADMA and AHUB
* XBAR : Crossbar for routing audio samples across various modules
* I2S : Inter-IC Sound Controller
* DMIC : Digital Microphone
* DSPK : Digital Speaker
Following is the summary of current series.
* Add YAML DT binding documentation for above mentioned modules.
* Helper function for ACIF programming is exposed for Tegra210 and later.
* Add ASoC driver components for each of the above modules.
* Build ACONNECT and ADMA drivers which are essential to realize audio
use case.
* Add DT entries for above components for Tegra210, Tegra186 and
Tegra194.
As per the suggestion in [0] audio graph based sound card support
is pushed in a separate series.
[0] https://lkml.org/lkml/2020/6/27/4
Changelog
=========
v4 -> v5
--------
* Common changes
- simple-card driver changes are dropped. Changes are migrated to audio
graph card and are moved to a separate series as suggested.
- '#sound-dai-cells' property is not needed for planned audio graph card
Hence dropped from documentation and related DT binding of component
drivers.
- CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to
represent DAI links using audio graph card. Similary DAIs are added in
AHUB driver to describe endpoints in audio crossbar. Routing is updated
to reflect the same in drivers.
v3 -> v4
--------
* [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210"
- Removed multiple examples and retained one example per doc
- Fixed as per inputs on the previous series
- Tested bindings with 'make dt_binding_check/dtbs_check'
* [2/23] "ASoC: tegra: Add support for CIF programming"
- No change
* Common changes (for patch [3/10] to [7/10])
- Mixer control overrides, for PCM parameters (rate, channel, bits),
in each driver are dropped.
- Updated routing as per DPCM usage
- Minor changes related to formatting
* New changes (patch [8/23] to [18/23] and patch [23/23])
- Based on discussions in following threads DPCM is used for Tegra Audio.
https://lkml.org/lkml/2020/2/20/91https://lkml.org/lkml/2020/4/30/519
- The simple-card driver is used for Tegra Audio and accordingly
some enhancements are made in simple-card and core drivers.
- Patch [8/23] to [18/23] are related to simple-card and core changes.
- Patch [23/23] adds sound card support to realize complete audio path.
This is based on simple-card driver with proposed enhancements.
- Re-ordered patches depending on above
v2 -> v3
--------
* [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB
- Updated licence
- Removed redundancy w.r.t items/const/enum
- Added constraints wherever needed with "pattern" property
* [2/10] "ASoC: tegra: add support for CIF programming"
- Removed tegra_cif.c
- Instead added inline helper function in tegra_cif.h
* common changes (for patch [3/10] to [7/10])
- Replace LATE system calls with Normal sleep
- Remove explicit RPM suspend in driver remove() call
- Use devm_kzalloc() instead of devm_kcalloc() for single element
- Replace 'ret' with 'err' for better reading
- Consistent error printing style across drivers
- Minor formating fixes
* [8/10] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later"
(New patch)
- Enables ACONNECT and AHUB components. With this AHUB and components are
registered with ASoC core.
v1 -> v2
--------
* [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB"
- no changes
* [2/9] "ASoC: tegra: add support for CIF programming"
- removed CIF programming changes for legacy chips.
- this patch now exposes helper function for CIF programming,
which can be used on Tegra210 later.
- later tegra_cif.c can be extended for legacy chips as well.
- updated commit message accordingly
* [3/9] "ASoC: tegra: add Tegra210 based DMIC driver"
- removed unnecessary initialization of 'ret' in probe()
* [4/9] "ASoC: tegra: add Tegra210 based I2S driver"
- removed unnecessary initialization of 'ret' in probe()
- fixed indentation
- added consistent bracing for if-else clauses
- updated 'rx_fifo_th' type to 'unsigned int'
- used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h
* [5/9] "ASoC: tegra: add Tegra210 based AHUB driver"
- used of_device_get_match_data() to get 'soc_data' and removed
explicit of_match_device()
- used devm_platform_ioremap_resource() and removed explicit
platform_get_resource()
- fixed indentation for devm_snd_soc_register_component()
- updated commit message
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [6/9] "ASoC: tegra: add Tegra186 based DSPK driver"
- removed unnecessary initialization of 'ret' in probe()
- updated 'max_th' to 'unsigned int'
- shortened lengthy macro names to avoid wrapping in
tegra186_dspk_wr_reg() and to be consistent
* [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver"
- used of_device_get_match_data() and removed explicit of_match_device()
- used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h
- updated commit message to reflect compatible binding for Tegra186 and
Tegra194.
* [8/9] "arm64: tegra: add AHUB components for few Tegra chips"
- no change
* [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips"
- no change
* common changes for patch [3/9] to [7/9]
- sorted headers in alphabetical order
- moved MODULE_DEVICE_TABLE() right below *_of_match table
- removed macro DRV_NAME
- removed explicit 'owner' field from platform_driver structure
- added 'const' to snd_soc_dai_ops structure
Sameer Pujar (11):
ASoC: dt-bindings: tegra: Add DT bindings for Tegra210
ASoC: tegra: Add support for CIF programming
ASoC: tegra: Add Tegra210 based DMIC driver
ASoC: tegra: Add Tegra210 based I2S driver
ASoC: tegra: Add Tegra210 based AHUB driver
ASoC: tegra: Add Tegra186 based DSPK driver
ASoC: tegra: Add Tegra210 based ADMAIF driver
arm64: defconfig: Build AHUB component drivers
arm64: defconfig: Build ADMA and ACONNECT driver
arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano
arm64: tegra: Add DT binding for AHUB components
.../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++
.../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++
.../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++
.../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++
arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++-
arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++-
arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 +
arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++
arch/arm64/configs/defconfig | 8 +
sound/soc/tegra/Kconfig | 56 ++
sound/soc/tegra/Makefile | 10 +
sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++
sound/soc/tegra/tegra186_dspk.h | 70 ++
sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++
sound/soc/tegra/tegra210_admaif.h | 162 ++++
sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++
sound/soc/tegra/tegra210_ahub.h | 127 ++++
sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++
sound/soc/tegra/tegra210_dmic.h | 82 +++
sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++
sound/soc/tegra/tegra210_i2s.h | 126 ++++
sound/soc/tegra/tegra_cif.h | 65 ++
sound/soc/tegra/tegra_pcm.c | 235 +++++-
sound/soc/tegra/tegra_pcm.h | 21 +-
25 files changed, 5251 insertions(+), 4 deletions(-)
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml
create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml
create mode 100644 sound/soc/tegra/tegra186_dspk.c
create mode 100644 sound/soc/tegra/tegra186_dspk.h
create mode 100644 sound/soc/tegra/tegra210_admaif.c
create mode 100644 sound/soc/tegra/tegra210_admaif.h
create mode 100644 sound/soc/tegra/tegra210_ahub.c
create mode 100644 sound/soc/tegra/tegra210_ahub.h
create mode 100644 sound/soc/tegra/tegra210_dmic.c
create mode 100644 sound/soc/tegra/tegra210_dmic.h
create mode 100644 sound/soc/tegra/tegra210_i2s.c
create mode 100644 sound/soc/tegra/tegra210_i2s.h
create mode 100644 sound/soc/tegra/tegra_cif.h
--
2.7.4
The Digital Speaker Controller (DSPK) converts the multi-bit Pulse Code
Modulation (PCM) audio input to oversampled 1-bit Pulse Density Modulation
(PDM) output. From the signal flow perpsective, the DSPK can be viewed as
a PDM transmitter that up-samples the input to the desired sampling rate
by interpolation then converts the oversampled PCM input to the desired
1-bit output via Delta Sigma Modulation (DSM).
This patch registers DSPK component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DSPK interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DSPK devices can be enabled in the DT via
"nvidia,tegra186-dspk" compatible binding. This driver can be used
on Tegra194 chip as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-7-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Audio Hub (AHUB) comprises a collection of hardware accelerators for
audio pre/post-processing and a programmable full crossbar (XBAR) for
routing audio data across these accelerators in time and in parallel.
AHUB supports multiple interfaces to I2S, DSPK, DMIC etc., XBAR is a
switch used to configure or modify audio routing between HW accelerators
present inside AHUB.
This patch registers AHUB component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes AHUB interfaces, which can be used to connect different
components in the ASoC layer. Currently the driver takes care of XBAR
programming to allow audio data flow through various clients of the AHUB.
Makefile and Kconfig support is added to allow to build the driver. The
AHUB component can be enabled in the DT via below compatible bindings.
- "nvidia,tegra210-ahub" for Tegra210
- "nvidia,tegra186-ahub" for Tegra186 and Tegra194
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-6-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Inter-IC Sound (I2S) controller implements full-duplex, bi-directional
and single direction point to point serial interface. It can interface
with I2S compatible devices. Tegra I2S controller can operate as both
master and slave.
This patch registers I2S controller with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes I2S interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The I2S devices can be enabled in the DT via
"nvidia,tegra210-i2s" compatible binding.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-5-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Digital MIC (DMIC) Controller is used to interface with Pulse Density
Modulation (PDM) input devices. The DMIC controller implements a converter
to convert PDM signals to Pulse Code Modulation (PCM) signals. From signal
flow perspective, the DMIC can be viewed as a PDM receiver.
This patch registers DMIC component with ASoC framework. The component
driver exposes DAPM widgets, routes and kcontrols for the device. The DAI
driver exposes DMIC interfaces, which can be used to connect different
components in the ASoC layer. Makefile and Kconfig support is added to
allow to build the driver. The DMIC devices can be enabled in the DT via
"nvidia,tegra210-dmic" compatible string. This driver can be used for
Tegra186 and Tegra194 chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1595134890-16470-4-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Audio Client Interface (CIF) is a proprietary interface employed to route
audio samples through Audio Hub (AHUB) components by inter connecting the
various modules.
This patch exports an inline function tegra_set_cif() which can be used,
for now, to program CIF on Tegra210 and later Tegra generations. Later it
can be extended to include helpers for legacy chips as well.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Reviewed-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/1595134890-16470-3-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This configuration is for EHL with the RT5660 codec. RT5660
should use "10EC5660" ID instead of "INTC1027".
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200717211337.31956-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All drivers are now using .mute_stream.
Let's remove .digital_mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87h7u72dqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Follow the recent inclusive terminology guidelines and replace the
word "slave" in vmaster API. I chose the word "follower" at this time
since it seems fitting for the purpose.
Note that the word "master" is kept in API, since it refers rather to
audio master volume control.
Also, while we're at it, a typo in comments is corrected, too.
Link: https://lore.kernel.org/r/20200717154517.27599-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200719151705.59624-1-grandmaster@al2klimov.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f52 (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
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Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
No surprise here, just a few device-specific small fixes: two fixes
for USB LINE6 and one for USB-audio drivers wrt syzkaller fuzzer
issues, while the rest are all HD-audio Realtek quirks.
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Merge tag 'sound-5.8-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai:
"No surprise here, just a few device-specific small fixes: two fixes
for USB LINE6 and one for USB-audio drivers wrt syzkaller fuzzer
issues, while the rest are all HD-audio Realtek quirks"
* tag 'sound-5.8-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - fixup for yet another Intel reference board
ALSA: hda/realtek - Enable Speaker for ASUS UX563
ALSA: hda/realtek - Enable Speaker for ASUS UX533 and UX534
ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
ALSA: hda/realtek - change to suitable link model for ASUS platform
ALSA: usb-audio: Fix race against the error recovery URB submission
ALSA: line6: Sync the pending work cancel at disconnection
ALSA: line6: Perform sanity check for each URB creation
Some settings should set to default value after the calibration.
This patch also disables the 25MHz and 1MHz clock power when the jack unplugged.
The JD is triggered by JDH, therefore this patch removes JDL setting.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070228.28660-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the function q6adm_open(), q6adm_alloc_copp() doesn't return
NULL. Thus use IS_ERR() to validate the returned value instead
of IS_ERR_OR_NULL(). And delete the extra line.
Signed-off-by: Zhang Shengju <zhangshengju@cmss.chinamobile.com>
Signed-off-by: Tang Bin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200714112744.20560-1-tangbin@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The pin status of the widget was connected after the sound card registered.
The rt5682_headset_detect function will use the pin status of these two widgets
to decide the certain register setting on/off.
Therefore this patch disables the pin of these two widgets in the codec probe.
This patch could avoid the misjudgment.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200717070256.28712-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is used for both CPU and Codec.
For example, soc_pcm_prepare() / soc_pcm_hw_free() are caring
both CPU and Codec.
But soc_resume_deferred() / snd_soc_suspend() are not.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87ft9r2dqr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
-
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Link: https://lore.kernel.org/r/87eepb2dnq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f6 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 3ad796cbc3 ("ALSA: pcm: Use SG-buffer only when
direct DMA is available") also the modification commit 467fd0e82b
("ALSA: pcm: Fix build error on m68k and others").
Poking the DMA internal helper is a layer violation, so we should
avoid that. Meanwhile the actual bug has been addressed by the
Kconfig fix in commit dbed452a07 ("dma-pool: decouple DMA_REMAP from
DMA_COHERENT_POOL"), so we can live without this hack.
Link: https://lore.kernel.org/r/20200717064130.22957-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi,
this small series is preparation for a set of bugfix ASoC patches
addressing a memleak at module unload for the HDA codec wrapper.
Instead of duplicating HDA code in ASoC tree, I chose to export
more functionality from hda_codec.c so it can be (re)used in ASoC's
hdac_hda.c.
Full series:
https://github.com/thesofproject/linux/pull/2252
Takashi and Mark, feedback is welcome on how to best handle this
kind of series where I have dependent patches both in sound/pci/hda
and in ASoC. For this series, I'm sending the patches separately
and when/if first set is merged by Takashi, I'll route the ASoC
patches via our usually SOF set to Mark.
Kai Vehmanen (2):
ALSA: hda: export snd_hda_codec_cleanup_for_unbind()
ALSA: hda: fix snd_hda_codec_cleanup() documentation
include/sound/hda_codec.h | 2 ++
sound/pci/hda/hda_codec.c | 3 ++-
2 files changed, 4 insertions(+), 1 deletion(-)
--
2.27.0
Support hp and mic detection.
Add a parameter for asoc_simple_init_jack.
Shengjiu Wang (3):
ASoC: simple-card-utils: Support configure pin_name for
asoc_simple_init_jack
ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio
ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection
changes in v2:
- Add more comments in third commit
- Add Acked-by Nicolin.
.../bindings/sound/fsl-asoc-card.txt | 3 +
include/sound/simple_card_utils.h | 6 +-
sound/soc/fsl/Kconfig | 1 +
sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++-
sound/soc/generic/simple-card-utils.c | 7 +-
5 files changed, 86 insertions(+), 8 deletions(-)
--
2.27.0
Add missed return for calling soc_component_ret, otherwise the return
value is wrong.
Fixes: e2329eeba4 ("ASoC: soc-component: add soc_component_err()")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/1594876028-1845-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use asoc_simple_init_jack function from simple card to implement
the Headphone and Microphone detection.
Register notifier to disable Speaker when Headphone is plugged in
and enable Speaker when Headphone is unplugged.
Register notifier to disable Digital Microphone when Analog Microphone
is plugged in and enable DMIC when Analog Microphone is unplugged.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-4-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the pin_name is fixed in asoc_simple_init_jack, but some driver
may use a different pin_name. So add a new parameter in
asoc_simple_init_jack for configuring pin_name.
If this parameter is NULL, then the default pin_name is used.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1594822179-1849-2-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87pn95wiwa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87r1tlwiwe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Link: https://lore.kernel.org/r/87sge1wiwi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87tuyhwiwm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87v9ixwiwr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87wo3dwiwv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87y2ntwix0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87zh89wix5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/871rllxxhp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/873661xxhu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/874kqhxxhz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/875zaxxxi4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/878sftxxie.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87a709xxij.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87blkpxxip.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
For hdmi-codec, we need to update struct hdmi_codec_ops,
and all its users in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87d055xxj2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling "direction".
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
To prepare merging mute_stream()/digital_mute(),
this patch adds .no_capture_mute support to emulate .digital_mute().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87eeplxxj7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() will return -ENOTSUPP if driver doesn't
support mute.
In hdmi-codec case, hdmi_codec_digital_mute() will be used for it,
and each driver has .digital_mute() callback.
hdmi_codec_digital_mute() want to return -ENOTSUPP to follow it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fta1xxjc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Vinod Koul