Kai Vehmanen [Fri, 9 Dec 2022 10:18:22 +0000 (12:18 +0200)]
ALSA: hda/hdmi: fix stream-id config keep-alive for rt suspend
When the new style KAE keep-alive implementation is used on compatible
Intel hardware, the clocks are maintained when codec is in D3. The
generic code in hda_cleanup_all_streams() can however interfere with
generation of audio samples in this mode, by setting the stream and
channel ids to zero.
To get full benefit of the keepalive, set the new
no_stream_clean_at_suspend quirk bit on affected Intel hardware. When
this bit is set, stream cleanup is skipped in hda_call_codec_suspend().
Special handling is needed for the case when system goes to suspend. The
stream id programming can be lost in this case. This will also cause
codec->cvt_setups to be out of sync. Handle this by implementing custom
suspend/resume handlers. If keep-alive is active for any converter, set
the quirk flags no_stream_clean_at_suspend and forced_resume. Upon
resume, keepalive programming is restored if needed.
Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2") Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20221209101822.3893675-4-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kai Vehmanen [Fri, 9 Dec 2022 10:18:21 +0000 (12:18 +0200)]
ALSA: hda/hdmi: set default audio parameters for KAE silent-stream
If the stream-id is zero, the keep-alive (KAE) will only ensure clock is
generated, but no audio samples are sent over display link. This happens
before first real audio stream is played out to a newly connected
receiver.
Reuse the code in silent_stream_enable() to set up stream parameters
to sane defaults values, also when using the newer keep-alive flow.
Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2") Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Tested-by: Rodrigo Vivi <rodrigo.vivi@intel.com> Link: https://lore.kernel.org/r/20221209101822.3893675-3-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
The i915 display codec may not successfully transition to
normal audio streaming mode, if the stream id is programmed
while codec is actively transmitting data. This can happen
when silent stream is enabled in KAE mode.
Fix the issue by implementing a i915 specific programming
flow, where the silent streaming is temporarily stopped,
a small delay is applied to ensure display codec becomes
idle, and then proceed with reprogramming the stream ID.
Cezary Rojewski [Thu, 8 Dec 2022 14:26:35 +0000 (15:26 +0100)]
ALSA: hda: Error out if invalid stream is being setup
Scenario when snd_hdac_stream_setup_periods() receives an instance of
struct hdac_stream with neither ->substream nor ->cstream initialized is
invalid.
Simultaneously addresses "uninitialized symbol 'dmab'" error reported by
Smatch.
Takashi Iwai [Fri, 9 Dec 2022 08:50:13 +0000 (09:50 +0100)]
Merge tag 'asoc-v6.2-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.2
A few more updates for v6.2 which can hopefully go into a later pull
request, the bulk of these are fixes, minor cleanups or new board quirks
- the one big bit that isn't is support for getting diagnostic data out
of the Intel AVS firmwares.
Chancel Liu [Mon, 28 Nov 2022 06:09:50 +0000 (14:09 +0800)]
ASoC: soc-pcm.c: Clear DAIs parameters after stream_active is updated
DAIs parameters should be cleared if there's no active stream. Before,
we implemented it in soc_pcm_hw_free() by detecting stream_active. If
the running stream is the last active stream, we're going to clear
parameters.
However it will cause DAIs parameters never be cleared if there're
more than one stream. For example, we have stream1 and stream2 about
to stop. stream2 executes soc_pcm_hw_free() before stream1 executes
soc_pcm_close(). At the moment, stream2 should clear DAIs parameters.
Since stream_active is not yet updated by stream1 in soc_pcm_close(),
stream2 will not clear DAIs parameters. In result both stream1 and
stream2 don't clear the parameters.
This patch moves DAIs parameters cleanup after stream_active is
updated.
Mark Brown [Wed, 7 Dec 2022 17:24:46 +0000 (17:24 +0000)]
ASoC: Intel: Skylake: Topology and shutdown fixes
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Even though skylake-driver is going to be replaced by the avs-driver,
the goal is to keep it functional on all the configurations it supports
until its EOL. When comparing chrome trees against upstream
skylake-driver, couple fixes pop up that are not part of upstream
repository. These fixes are backed up by real bugs (issue trackers),
address real problems. There is no reason for them to stay in the
internal tree.
Patches 1-4 combined together address issue where the driver updates the
presumably static audio format descriptions coming from the topology
files through its "fixup" functions. As long as given audio format is
used by a single path, nothing collides and any updates are harmless.
However, when multiple paths e.g.: DMIC and HDMI1 utilize the same audio
format descriptor, any updates caused by the opening of the first path,
may cause the second to fail.
The 5th change from the set fixes driver hang sporadically occurring
during shutdown procedure. Once HDAudio links are powered down along
with the AudioDSP, the hang stops reproducing.
The last change helps survive in environments with limited/fragmented
memory. While the BDL is small already, other buffers can be allocated
using scatter-gather. This basically aligns the code with what the
avs-driver does.
Wang Yufen [Mon, 5 Dec 2022 09:56:28 +0000 (17:56 +0800)]
ASoC: mediatek: mt8183: fix refcount leak in mt8183_mt6358_ts3a227_max98357_dev_probe()
The node returned by of_parse_phandle() with refcount incremented,
of_node_put() needs be called when finish using it. So add it in the
error path in mt8183_mt6358_ts3a227_max98357_dev_probe().
ye xingchen [Mon, 5 Dec 2022 11:43:47 +0000 (19:43 +0800)]
ASoC: imx-audmux: use sysfs_emit() to instead of scnprintf()
Follow the advice of the Documentation/filesystems/sysfs.rst and show()
should only use sysfs_emit() or sysfs_emit_at() when formatting the
value to be returned to user space.
Wang Yufen [Mon, 5 Dec 2022 08:15:27 +0000 (16:15 +0800)]
ASoC: audio-graph-card: fix refcount leak of cpu_ep in __graph_for_each_link()
The of_get_next_child() returns a node with refcount incremented, and
decrements the refcount of prev. So in the error path of the while loop,
of_node_put() needs be called for cpu_ep.
Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wang Yufen <wangyufen@huawei.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/1670228127-13835-1-git-send-email-wangyufen@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Wang Yufen [Mon, 5 Dec 2022 10:04:24 +0000 (18:04 +0800)]
ASoC: mediatek: mt8173-rt5650-rt5514: fix refcount leak in mt8173_rt5650_rt5514_dev_probe()
The node returned by of_parse_phandle() with refcount incremented,
of_node_put() needs be called when finish using it. So add it in the
error path in mt8173_rt5650_rt5514_dev_probe().
Cezary Rojewski [Mon, 5 Dec 2022 08:53:30 +0000 (09:53 +0100)]
ASoC: Intel: Skylake: Use SG allocation for SKL-based firmware load
Resign from ->alloc_dma_buf() and use snd_dma_alloc_pages() directly.
For data i.e.: base firmware binary transfer, make use of SG allocation
to better adapt to memory-limited environment. For BDL descriptor, given
its small size this is not required.
Cezary Rojewski [Mon, 5 Dec 2022 08:53:28 +0000 (09:53 +0100)]
ASoC: Intel: Skylake: Introduce single place for pipe-config selection
Provide a single location for pipe config selection where all fields
that have to be updated whenever ->pipe_config_idx changes can be
updated accordingly.
Jiao Zhou [Tue, 6 Dec 2022 18:53:11 +0000 (13:53 -0500)]
ALSA: hda/hdmi: Add HP Device 0x8711 to force connect list
HDMI audio is not working on the HP EliteDesk 800 G6 because the pin is
unconnected. This issue can be resolved by using the 'hdajackretask'
tool to override the unconnected pin to force it to connect.
Takashi Iwai [Tue, 6 Dec 2022 10:13:26 +0000 (11:13 +0100)]
Merge tag 'asoc-v6.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.2
This is a fairly sedate release for the core code, but there's been a
lot of driver work especially around the x86 platforms and device tree
updates:
- More cleanups of the DAPM code from Morimoto-san.
- Factoring out of mapping hw_params onto SoundWire configuration by
Charles Keepax.
- The ever ongoing overhauls of the Intel DSP code continue, including
support for loading libraries and probes with IPC4 on SOF.
- Support for more sample formats on JZ4740.
- Lots of device tree conversions and fixups.
- Support for Allwinner D1, a range of AMD and Intel systems, Mediatek
systems with multiple DMICs, Nuvoton NAU8318, NXP fsl_rpmsg and
i.MX93, Qualcomm AudioReach Enable, MFC and SAL, RealTek RT1318 and
Rockchip RK3588
There's more cross tree updates than usual, though all fairly minor:
- Some OMAP board file updates that were depedencies for removing their
providers in ASoC, as part of a wider effort removing the support for
the relevant OMAP platforms.
- A new I2C API required for updates to the new I2C probe API.
- A DRM update making use of a new API for fixing the capabilities
advertised via hdmi-codec.
Since this is being sent early I might send some more stuff if you've
not yet sent your pull request and there's more come in.
Mark Brown [Mon, 5 Dec 2022 19:01:57 +0000 (19:01 +0000)]
ASoC: Intel: avs: Data probing and fw logging
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
The patchset focuses on debug functionality for the avs-driver.
Two major blocks are covered here: data probing and AudioDSP firmware
logging. Both are configured and controlled through debugfs.
Data probing is a AudioDSP debug functionality which allows for
gathering the actual data that is being routed to or from a module.
Helps in debugging its processing capabilities - navigate to a specific
module which may have caused a glitch within a pipeline (set of modules
bound together).
First few allow for assigning compress stream to a HDAudio stream, what
is currently limited to pcm substreams only. These patches were already
present on this list and reviewed in the past [1].
The next few tidy existing debug-related code up so it's ready for
addition of new functionalities and make it clear which part of the avs
is debug related and which is not. These also simplify the existing
locking around the trace fifo.
Afterward, debug-related IPCs are defined along with stub soc-component
and compress DAI operations. Not much is done there as it's not a
standard PCM streaming scenario. Most code found in compress operations
is inherited from the HOST side of HDAudio streaming found in pcm.c
file of the driver.
Finally, a debugfs file operations are defined. These facilitate
connecting to DSP modules from which the data shall be gathered as well
as control and configuration of firmware logging. Additionally, entries
are added to allow for dumping snapshots of key memory windows.
Yang Yingliang [Mon, 5 Dec 2022 14:37:21 +0000 (22:37 +0800)]
ASoC: sof_es8336: fix possible use-after-free in sof_es8336_remove()
sof_es8336_remove() calls cancel_delayed_work(). However, that
function does not wait until the work function finishes. This
means that the callback function may still be running after
the driver's remove function has finished, which would result
in a use-after-free.
Fix by calling cancel_delayed_work_sync(), which ensures that
the work is properly cancelled, no longer running, and unable
to re-schedule itself.
Fixes: 89cdb224f2ab ("ASoC: sof_es8336: reduce pop noise on speaker") Signed-off-by: Yang Yingliang <yangyingliang@huawei.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20221205143721.3988988-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Philipp Jungkamp [Mon, 5 Dec 2022 16:37:13 +0000 (17:37 +0100)]
ALSA: patch_realtek: Fix Dell Inspiron Plus 16
The Dell Inspiron Plus 16, in both laptop and 2in1 form factor, has top
speakers connected on NID 0x17, which the codec reports as unconnected.
These speakers should be connected to the DAC on NID 0x03.
Krzysztof Kozlowski [Sun, 4 Dec 2022 11:36:21 +0000 (12:36 +0100)]
ASoC: dt-bindings: maxim,max98504: Convert to DT schema
Convert the Maxim Integrated MAX98504 amplifier bindings to DT schema.
Few properties are made optional:
1. interrupts: current Linux driver implementation does not use them,
2. supplies: on some boards these might be wired to battery, for which
no regulator is provided.
Krzysztof Kozlowski [Sat, 3 Dec 2022 16:04:42 +0000 (17:04 +0100)]
ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
Convert the Maxim Integrated MAX98357A/MAX98360A amplifier bindings to
DT schema. Add missing properties ('#sound-dai-cells' and
'sound-name-prefix' from common DAI properties).
Krzysztof Kozlowski [Sat, 3 Dec 2022 16:04:41 +0000 (17:04 +0100)]
ASoC: dt-bindings: Reference common DAI properties
Reference in all sound components which have '#sound-dai-cells' the
dai-common.yaml schema, which allows to use 'sound-name-prefix'
property.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Tested-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com> Acked-by: Nicolas Frattaroli <frattaroli.nicolas@gmail.com> Link: https://lore.kernel.org/r/20221203160442.69594-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
Krzysztof Kozlowski [Sat, 3 Dec 2022 16:04:40 +0000 (17:04 +0100)]
ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
Rename name-prefix.yaml into common DAI schema and document
'#sound-dai-cells' for completeness. The '#sound-dai-cells' cannot be
really constrained, as there are users with value of 0, 1 and 2, but at
least it brings definition to one common place.
Colin Ian King [Fri, 2 Dec 2022 17:14:50 +0000 (17:14 +0000)]
ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
Don't populate the read-only arrays capture_reg_H and capture_reg_L
on the stack but instead make them static const. Also makes the
object code a little smaller.
Colin Ian King [Fri, 2 Dec 2022 16:41:56 +0000 (16:41 +0000)]
ASoC: uniphier: aio-core: Make some read-only arrays static const
Don't populate the read-only arrays slotsel_2ch, slotsel_multi, v_pll
and v_div on the stack but instead make them static const. Also makes
the object code a little smaller.
Ajye Huang [Mon, 5 Dec 2022 12:06:48 +0000 (17:36 +0530)]
ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
The Skyrim project and Whiterun met error when DSP
loading during device boot.
Ex, error in kernel log,
ERR kernel: [ 16.124537] snd_sof_amd_rembrandt
0000:04:00.5: PSP validation failed.
Use the snd_sof_dsp_read_poll_timeout function to successfully
read the FW_QUALIFIER register
Cezary Rojewski [Fri, 2 Dec 2022 15:28:40 +0000 (16:28 +0100)]
ASoC: Intel: avs: Allow for dumping FW_REGS area
SRAM0 window begins with a block of memory, usually of size PAGE_SIZE,
dedicated to the base firmware registers. When debugging firmware, it is
desirable to be able to dump them at will.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:39 +0000 (16:28 +0100)]
ASoC: Intel: avs: Gather remaining logs on strace_release()
When user closes the tracer, some logs may still remain in the tail of
the buffer as firmware sends LOG_BUFFER_STATUS notification only when
certain threshold of data is reached. Add whatever is left to already
gathered logs so no information is lost.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:38 +0000 (16:28 +0100)]
ASoC: Intel: avs: Probing and firmware tracing over debugfs
Define debugfs subdirectory delegated for IPC communication with DSP.
Input format: uint,uint,(...) which are later translated into DWORDS
sequence and further into instances of struct of interest given the IPC
type.
For Extractor probes, following have been enabled:
- PROBE_POINT_ADD (echo <..> probe_points)
- PROBE_POINT_REMOVE (echo <..> probe_points_remove)
- PROBE_POINT_INFO (cat probe_points)
Cezary Rojewski [Fri, 2 Dec 2022 15:28:36 +0000 (16:28 +0100)]
ASoC: Intel: avs: Data probing soc-component
Define stub component for data probing. Stub as most operations from
standard PCM case do not apply here. Specific bits are CPU DAIs and
compress_ops. FE DAIs can link against these new CPU DAI to create new
compress devices.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:35 +0000 (16:28 +0100)]
ASoC: Intel: avs: Probe compress operations
Add compress operations handlers for data extraction through probes. A
single HDAudio stream is enlisted for said purpose. Operations follow
same protocol as for standard PCM streaming on HOST side.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:34 +0000 (16:28 +0100)]
ASoC: Intel: avs: Add data probing requests
Data probing is a cAVS firmware functionality that allows for data
extraction and injection directly from or to DMA stream. To support it,
new functions and types are added. These facilitate communication
with the firmware.
Total of eight IPCs:
- probe module initialization and cleanup
- addition and removal of probe points
- addition and removal of injection DMAs
- dumping list of currently connected probe points or enlisted DMAs
Cezary Rojewski [Fri, 2 Dec 2022 15:28:33 +0000 (16:28 +0100)]
ASoC: Intel: avs: Drop usage of debug members in non-debug code
Switch to debug-context aware wrappers instead of accessing debug
members directly allowing for readable separation of debug and non-debug
related code. Duplicates are removed along the way.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:32 +0000 (16:28 +0100)]
ASoC: Intel: avs: Make enable_logs() dependent on DEBUG_FS
Without debug filesystem present, this code is redundant.
Operations: log_buffer_status and log_buffer_offset are left as is as
EXCEPTION_CAUGHT and even unexpected LOG_BUFFER_STATUS notifications may
occur without user ever touching debugfs.
Debug-related fields and log-dumping are useful when debugfs is enabled.
Define them under CONFIG_DEBUG_FS and provide stubs when the config is
disabled so that the code that makes use of these needs not to be
complicated unnecessarily.
Members that are duplicated by this patch will be removed by the follow
up changes.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:28 +0000 (16:28 +0100)]
ALSA: hda: Interrupt servicing and BDL setup for compress streams
Account for compress streams when receiving and servicing buffer
completed interrupts. In case of compress stream enlisting hdac_stream
for data transfer, setup BDL entries much like it is the case for PCM
streams.
Cezary Rojewski [Fri, 2 Dec 2022 15:28:26 +0000 (16:28 +0100)]
ALSA: hda: Allow for compress stream to hdac_ext_stream assignment
Currently only PCM streams can enlist hdac_stream for their data
transfer. Add cstream field to hdac_ext_stream to expose possibility of
compress stream assignment in place of PCM one.
Limited to HOST-type only as there no other users on the horizon.
Takashi Iwai [Mon, 5 Dec 2022 13:21:24 +0000 (14:21 +0100)]
ALSA: usb-audio: Workaround for XRUN at prepare
Under certain situations (typically in the implicit feedback mode),
USB-audio driver starts a playback stream already at PCM prepare call
even before the actual PCM trigger-START call. For implicit feedback
mode, this effectively starts two streams for data and sync
endpoints, and if a coupled sync stream gets XRUN at this point, it
results in an error -EPIPE.
The problem is that currently we return -EPIPE error as is from the
prepare. Then application tries to recover again via the prepare
call, but it'll fail again because the sync-stop is missing. The
sync-stop is missing because it's an internal trigger call (hence the
PCM core isn't involved).
Since we'll need to re-issue the prepare in anyway when trapped into
this pitfall, this patch attempts to address it in a bit different
way; namely, the driver tries to prepare once again after syncing the
stop manually by itself -- so applications don't see the internal
error. At the second failure, we report the error as is, but this
shouldn't happen in normal situations.
Takashi Iwai [Mon, 5 Dec 2022 13:21:23 +0000 (14:21 +0100)]
ALSA: pcm: Handle XRUN at trigger START
When the driver returns -EPIPE for indicating an XRUN already at PCM
trigger START, we should treat properly and set it to the XRUN state.
Otherwise the state is missing and the application would try to issue
trigger again without knowing that it's in an error state.
This is just for a theoretical bug, and it won't happen in most
cases.
Takashi Iwai [Mon, 5 Dec 2022 13:21:22 +0000 (14:21 +0100)]
ALSA: pcm: Set missing stop_operating flag at undoing trigger start
When a PCM trigger-start fails at snd_pcm_do_start(), PCM core tries
to undo the action at snd_pcm_undo_start() by issuing the trigger STOP
manually. At that point, we forgot to set the stop_operating flag,
hence the sync-stop won't be issued at the next prepare or other
calls.
This patch adds the missing stop_operating flag at
snd_pcm_undo_start().
Mark Brown [Sun, 4 Dec 2022 17:01:50 +0000 (17:01 +0000)]
ASoC/tda998x: Fix reporting of nonexistent capture streams
Merge series from Mark Brown <broonie@kernel.org>:
The recently added pcm-test selftest has pointed out that systems with
the tda998x driver end up advertising that they support capture when in
reality as far as I can see the tda998x devices are transmit only. The
DAIs registered through hdmi-codec are bidirectional, meaning that for
I2S systems when combined with a typical bidrectional CPU DAI the
overall capability of the PCM is bidirectional. In most cases the I2S
links will clock OK but no useful audio will be returned which isn't so
bad but we should still not advertise the useless capability, and some
systems may notice problems for example due to pinmux management.
This is happening due to the hdmi-codec helpers not providing any
mechanism for indicating unidirectional audio so add one and use it in
the tda998x driver. It is likely other hdmi-codec users are also
affected but I don't have those systems to hand.
Mark Brown (2):
ASoC: hdmi-codec: Allow playback and capture to be disabled
drm: tda99x: Don't advertise non-existent capture support
Mark Brown [Wed, 30 Nov 2022 18:46:43 +0000 (18:46 +0000)]
ASoC: hdmi-codec: Allow playback and capture to be disabled
Currently the hdmi-codec driver always registers both playback and capture
capabilities but for most systems there's no actual capture capability,
usually HDMI is transmit only. Provide platform data which allows the users
to indicate what is supported so that we don't end up advertising things
to userspace that we can't actually support.
In order to avoid breaking existing users the flags in platform data are
a bit awkward and specify what should be disabled rather than doing the
perhaps more expected thing and defaulting to not supporting capture.
Mark Brown [Thu, 1 Dec 2022 17:07:45 +0000 (17:07 +0000)]
kselftest/alsa: Add more coverage of sample rates and channel counts
Now that we can skip unsupported configurations add some more test cases
using that, cover 8kHz, 44.1kHz and 96kHz plus 8kHz mono and 48kHz 6
channel.
44.1kHz is a different clock base to the existing 48kHz tests and may
therefore show problems with the clock configuration if only 8kHz based
rates are really available (or help diagnose if bad clocking is due to
only 44.1kHz based rates being supported). 8kHz mono and 48Hz 6 channel
are real world formats and should show if clocking does not account for
channel count properly.
Mark Brown [Thu, 1 Dec 2022 17:07:44 +0000 (17:07 +0000)]
kselftest/alsa: Provide more meaningful names for tests
Rather than just numbering the tests try to provide semi descriptive names
for what the tests are trying to cover. This also has the advantage of
meaning we can add more tests without having to keep the list of tests
ordered by existing number which should make it easier to understand what
we're testing and why.
Mark Brown [Thu, 1 Dec 2022 17:07:43 +0000 (17:07 +0000)]
kselftest/alsa: Don't any configuration in the sample config
The values in the one example configuration file we currently have are the
default values for the two tests we have so there's no need to actually set
them. Comment them out as examples, with a rename for the tests so that we
can update the tests in the code more easily.
Mark Brown [Thu, 1 Dec 2022 17:07:42 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested channels as skips
If constraint selection gives us a number of channels other than the one
that we asked for that isn't a failure, that is the device implementing
constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:41 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested sample rate as skips
If constraint selection gives us a sample rate other than the one that we
asked for that isn't a failure, that is the device implementing sample
rate constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:40 +0000 (17:07 +0000)]
kselftest/alsa: Refactor pcm-test to list the tests to run in a struct
In order to help make the list of tests a bit easier to maintain refactor
things so we pass the tests around as a struct with the parameters in,
enabling us to add new tests by adding to a table with comments saying
what each of the number are. We could also use named initializers if we get
more parameters.
David Rau [Mon, 21 Nov 2022 05:07:44 +0000 (05:07 +0000)]
ASoC: da7219: Fix pole orientation detection on OMTP headsets when playing music
The OMTP pin define headsets can be mis-detected as line out
instead of OMTP, causing obvious issues with audio quality.
This patch is to put increased resistances within
the device at a suitable point.
To solve this issue better, the new mechanism setup
ground switches with conditional delay control
and these allow for more stabile detection process
to operate as intended. This conditional delay control
will not impact the hardware process
but use extra system resource.
This commit improves control of ground switches in the AAD logic.
Takashi Iwai [Wed, 30 Nov 2022 16:26:55 +0000 (17:26 +0100)]
Merge tag 'asoc-fix-v6.1-rc7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.1
Some more fixes for v6.1, some of these are very old and were originally
intended to get sent for v5.18 but got lost in the shuffle when there
was an issue with Linus not liking my branching strategy and I rebuilt
bits of my workflow. The ops changes have been validated by people
looking at real hardware and are how things getting dropped got noticed.
Takashi Sakamoto [Wed, 30 Nov 2022 14:33:13 +0000 (23:33 +0900)]
ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC
TC Applied Technologies (TCAT) produces TCD3070 as final DICE ASIC for
communication in IEEE 1394 bus for IEC 61883-1/6 protocol. As long as I
know, latter model of Focusrite Saffire Pro 40 is an application of the
ASIC and only in the market for consumers.
This patchset adds support for the device. The device has several
remarkable points.
1. No support for extended synchronization information section in TCAT
general protocol. The value of GLOBAL_EXTENDED_STATUS register is always
zero. Additionally, NOTIFY_EXT_STATUS message is never emitted.
2. No support for TCAT protocol extension. Hard coding is required for
format of CIP payload.
3. During several seconds after changing sampling rate, the block to
process PCM frames is under disfunction. When starting packet streaming
during the state, the block is never function till configuring different
sampling rate and several seconds.
This commit adds support for the device. The item 1 and 2 can be
adaptable, while item 3 is not. It's not preferable that user process
is forced to sleep during the disfunction in the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS or SNDRV_PCM_IOCTL_PREPARE request. It's
inconvenient but let user configure preferable sampling rate in advance
of starting PCM substream.
The content of configuration ROM in the device I used is available at:
* https://github.com/takaswie/am-config-roms/
I note that any mixer control operation is implemented by unique
transaction. The frame of request consists of 16 bytes header followed
by payload.
header (4 quadlets):
1st: the type of request, prefixed with 0x8000
2nd: counter at 2 bytes in MSB side, the length of data at 2 bytes in LSB
side
3rd: parameter 0
4th: parameter 1
payload (variable length if need):
5th-: data according to parameters
The request frame is sent by block write request to 0x'ffff'e040'01c0.
The frame of response is similar to the frame of request, but it is
header only, thus fixed to 16 bytes. The response frame is sent to the
address which is registered by lock transaction to 0x'ffff'e040'0008.
If the operation results in batch of data, the 2nd quadlet of header
includes the length of data like request. The data is itself readable
by read block request to 0x'ffff'e040'0030, which includes both
header and payload for data, thus the length to read should be the
length of data plus 16 bytes for header
The actual value of request, parameter 0, parameter 1, and data is
unclear yet.
Takashi Sakamoto [Wed, 30 Nov 2022 13:06:04 +0000 (22:06 +0900)]
ALSA: dice: fix regression for Lexicon I-ONIX FW810S
For Lexicon I-ONIX FW810S, the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS can returns -ETIMEDOUT. This is a regression due
to the commit 41319eb56e19 ("ALSA: dice: wait just for
NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation"). The device
does not emit NOTIFY_CLOCK_ACCEPTED notification when accepting
GLOBAL_CLOCK_SELECT operation with the same parameters as current ones.
This commit fixes the regression. When receiving no notification, return
-ETIMEDOUT as long as operating for any change.
Fixes: 41319eb56e19 ("ALSA: dice: wait just for NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20221130130604.29774-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai@suse.de>
Artem Lukyanov [Wed, 30 Nov 2022 08:52:47 +0000 (11:52 +0300)]
ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table
This model requires an additional detection quirk to enable the
internal microphone - BIOS doesn't seem to support AcpDmicConnected
(nothing in acpidump output).
Mark Brown [Tue, 29 Nov 2022 19:29:05 +0000 (19:29 +0000)]
ASoC: Intel: avs: rt5682: Refactor jack handling
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Leftover from recent series [1].
Following changes are proposed for the rt5682 sound card driver:
1) Move jack unassignment from platform_device->remove() to
dai_link->exit(). This is done to make jack init and deinit flows
symmetric
2) Remove platform_device->remove() function
3) Simplify card->suspend_pre() and card->resume_post() by making use of
snd_soc_card_get_codec_dai() helper
Amadeusz Sławiński [Tue, 29 Nov 2022 18:07:38 +0000 (19:07 +0100)]
ASoC: Intel: avs: rt5682: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
ASoC: pcm512x: Fix PM disable depth imbalance in pcm512x_probe
The pm_runtime_enable will increase power disable depth. Thus
a pairing decrement is needed on the error handling path to
keep it balanced according to context. We fix it by going to
err_pm instead of err_clk.
Fixes:f086ba9d5389c ("ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL")
Mark Brown [Tue, 29 Nov 2022 16:56:44 +0000 (16:56 +0000)]
ASoC: Intel: avs: Refactor jack handling
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
For all the boards included in this patchset, a similar set of changes
is proposed:
1) Move jack unassignment from platform_device->remove() to
dai_link->exit(). This is done to make jack init and deinit flows
symmetric
2) Remove platform_device->remove() function
3) Simplify card->suspend_pre() and card->resume_post() by making use of
snd_soc_card_get_codec_dai() helper
While bdw_rt286 board - which is utilized by the catpt-driver - is
definitely not part of "avs", same treatment applies. And thus decided
to make it part of this series instead of sending it separately.
Jaroslav Kysela [Tue, 29 Nov 2022 08:53:06 +0000 (09:53 +0100)]
selftests: alsa - move shared library configuration code to conf.c
The minimal alsa-lib configuration code is similar in both mixer
and pcm tests. Move this code to the shared conf.c source file.
Also, fix the build rules inspired by rseq tests. Build libatest.so
which is linked to the both test utilities dynamically.
Also, set the TEST_FILES variable for lib.mk.
Cc: linux-kselftest@vger.kernel.org Cc: Shuah Khan <shuah@kernel.org> Reported-by: Mark Brown <broonie@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Tested-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20221129085306.2345763-1-perex@perex.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
John Keeping [Tue, 29 Nov 2022 13:00:59 +0000 (13:00 +0000)]
ALSA: usb-audio: Add quirk for Tascam Model 12
Tascam's Model 12 is a mixer which can also operate as a USB audio
interface. The audio interface uses explicit feedback but it seems that
it does not correctly handle missing isochronous frames.
When injecting an xrun (or doing anything else that pauses the playback
stream) the feedback rate climbs (for example, at 44,100Hz nominal, I
see a stable rate around 44,099 but xrun injection sees this peak at
around 44,135 in most cases) and glitches are heard in the audio stream
for several seconds - this is significantly worse than the single glitch
expected for an underrun.
While the stream does normally recover and the feedback rate returns to
a stable value, I have seen some occurrences where this does not happen
and the rate continues to increase while no audio is heard from the
output. I have not found a solid reproduction for this.
This misbehaviour can be avoided by totally resetting the stream state
by switching the interface to alt 0 and back before restarting the
playback stream.
Add a new quirk flag which forces the endpoint and interface to be
reconfigured whenever the stream is stopped, and use this for the Tascam
Model 12.
Separate interfaces are used for the playback and capture endpoints, so
resetting the playback interface here will not affect the capture stream
if it is running. While there are two endpoints on the interface,
these are the OUT data endpoint and the IN explicit feedback endpoint
corresponding to it and these are always stopped and started together.
Amadeusz Sławiński [Fri, 25 Nov 2022 18:40:30 +0000 (19:40 +0100)]
ASoC: Intel: avs: rt298: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Amadeusz Sławiński [Fri, 25 Nov 2022 18:40:28 +0000 (19:40 +0100)]
ASoC: Intel: avs: rt286: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Amadeusz Sławiński [Fri, 25 Nov 2022 18:40:26 +0000 (19:40 +0100)]
ASoC: Intel: avs: rt274: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Amadeusz Sławiński [Fri, 25 Nov 2022 18:40:24 +0000 (19:40 +0100)]
ASoC: Intel: avs: nau8825: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
Amadeusz Sławiński [Fri, 25 Nov 2022 18:40:23 +0000 (19:40 +0100)]
ASoC: Intel: avs: da7219: Refactor jack handling
Use link->exit() rather than pdev->remove() to unassign jack during card
unbind procedure so codec link initialization and exit procedures are
symmetrical.
Also, there is no need to perform search for codec dai in suspend_pre()
and resume_post() ourselves. Use snd_soc_card_get_codec_dai() instead.
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