Peter Ujfalusi [Tue, 9 Apr 2024 08:38:10 +0000 (11:38 +0300)]
ALSA: pci: hda: hda_controller: Add support for use_pio_for_commands mode
Set the use_pio_for_commands flag in case AZX_DCAPS_PIO_COMMANDS quirk is
enabled.
When the PIO command mode is used we can re-use the existing
azx_single_send_cmd() / azx_single_get_response() functions safely as the
CORB DMA is not going to be enabled in snd_hdac_bus_init_cmd_io().
Peter Ujfalusi [Tue, 9 Apr 2024 08:38:09 +0000 (11:38 +0300)]
ALSA: hda: hdac_controller: Implement support for use_pio_for_commands mode
In case the use_pio_for_commands flag is set we must not enable the
CORB DMA to make sure that it is not interfering with the immediate
command mode.
Convert the snd_hdac_bus_send_cmd/snd_hdac_bus_get_response as wrappers to
call either the PIO or CORB based command handling depending on the
use_pio_for_commands flag.
Peter Ujfalusi [Tue, 9 Apr 2024 08:38:08 +0000 (11:38 +0300)]
ALSA: hda: Introduce flags to force commands via PIO instead of CORB
Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to
be able to select PIO mode as alternative for CORB based command sending
while retaining the RIRB functionality to receive unsolicited responses.
This mode differs from the azx single_cmd mode when RIRB is disabled.
The mixed mode is needed on Lunar Lake family because it is recommended to
use Immediate Command Response (PIO mode) instead of CORB for HDA commands.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:37:56 +0000 (05:07 +1030)]
ALSA: scarlett2: Add autogain target controls
The Scarlett 4th Gen and Vocaster interfaces allow the autogain target
dBFS value(s) to be configured. Add Mean and Peak Target controls for
4th Gen, and a Hot Target control for Vocaster.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:37:12 +0000 (05:07 +1030)]
ALSA: scarlett2: Add DSP controls
Add filter and compressor DSP controls for the Vocaster interfaces.
Mark scarlett2_notify_input_dsp() as __always_unused until it gets
used when the Vocaster callback function array is added.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:36:23 +0000 (05:06 +1030)]
ALSA: scarlett2: Add input mute controls
Add controls for the input mute switches that the Vocaster interfaces
have. Mark scarlett2_notify_input_mute() as __always_unused until it
gets used when the Vocaster callback function array is added.
The 4th Gen Scarlett interfaces added software-controllable input gain
along with channel select, channel link, auto-gain, and "safe" mode.
Vocaster has software-controllable input gain and auto-gain but not
channel select, channel link, or safe mode.
Add a device info field safe_input_count to indicate how many channels
have a safe mode control, and use the presence of the input select and
input link switch configuration parameters to determine if those
controls should be created.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:35:15 +0000 (05:05 +1030)]
ALSA: scarlett2: Add pbuf field to struct scarlett2_config
scarlett2_usb_set_config() was using size = 0 as a signal to use the
parameter buffer. Replace that with an explicit indication (pbuf = 1),
as the upcoming Vocaster support has a config item written via the
parameter buffer with size = 1 rather than the implicit size of 8.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:34:14 +0000 (05:04 +1030)]
ALSA: scarlett2: Implement handling of the ACK notification
After scarlett2_usb() sends a command, it seems that we should wait
for an ACK before attempting to read the response. Not doing that
didn't seem necessary previously but seems to be causing occasional
issues with 4th Gen devices.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:33:14 +0000 (05:03 +1030)]
ALSA: scarlett2: Move initialisation code lower in the source
So that more forward declarations won't be required when we add
handling of the ACK notification, move the initialisation functions to
after the notification functions.
Both drivers provide both sample_new and sample_free, and it makes no
sense to pretend that they could not. In fact, load_data() would already
crash if sample_new was null. So remove the remaining null checks.
Contrary to that, the emu10k1 driver actually has a null sample_reset,
though I'm not convinced that this inconsistency is justified.
ALSA: emu10k1: shrink blank space in front of wavetable samples
There is no need for it to be 32 samples - 3 will do just fine (which is
the interpolator's epsilon). The old size was presumably meant to
compensate for the cache's presence, but we're now handling that
properly.
ALSA: emu10k1: fix wavetable playback position and caching, take 2
Compensate for the cache lag of 64 frames, and actually populate the
cache. Without these, the playback would start with garbage (which
would be (mostly?) masqueraded by the note's attack phase).
Note that we set the starting address only 61 frames ahead, to
compensate for the interpolator's epsilon. Unlike for PCM playback, we
don't even need to manually silence-fill the first frames in the cache,
because we insert some silence in front of each sample anyway.
A challenge are extremely short samples with a loop end below the cache
size, because a) we'd have to wrap the current address to be within the
loop and b) automatic pre-filling of the cache with the right data does
not work in this case.
We could pre-fill the cache manually, but that's slow, requires
additional code for each sample width, and is made even more complex by
the driver's virtual address space having no contiguous mapping for the
CPU.
We could have the engine fill the cache piece-wise (which is really what
happens when playback is running), but that would also be complex, and
we'd need to wait for the engine to handle each piece, so it wouldn't be
that much faster than the manual fill.
For the case of requiring only one loop iteration prior to reaching the
cache size, we could leverage the engine's looping mechanism around
CCR_CACHELOOPFLAG, but this special case doesn't seem worth the
complexity.
So we just unroll the loop as far as necessary to be able to play back
the sample without any fiddling.
Pedantically, this would be incorrect for loop-until-release samples
with a low loop end which are released very quickly, but that would be
relatively harmless, is not a plausible use case in the first place, and
SoundFont sample mode 3 isn't actually implemented anyway (it's
conflated with mode 1, infinite looping).
ALSA: emu10k1: move patch loader assertions into low-level functions
Convert some checks in snd_emu10k1_sample_new() back into assertions (as
they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*,
2008-08-08)), and move them into the low-level memory access functions
they protect.
ALSA: emux: centralize & improve patch info validation
This does several closely related things:
- Move the code from the drivers into the SoundFont loader, which
de-duplicates it.
- Sort of explain the weird "recalculate address offset" feature. Note
that I don't think it actually makes any sense - the calling user
space code should do that. The background is certainly that the source
data (the SoundFont format) uses pointers into a single wave block
(and the API allows doing the same for on-board ROM), but the API
expects the wave data from user space to be pre-chopped into
individual patches anyway.
- Make sure that the specified offsets actually lie within the supplied
wave data. Note that we don't validate ROM offsets, so one can play
back anything within the sound card's address space.
- In load_guspatch(), don't call the sample_new callback anymore when
the patch size is zero, as was already the case in load_data(). The
callbacks would instantly return in that case anyway; these checks are
now removed.
ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support
This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in
the GUS patch loader. It has not worked on emu10k1 since before ALSA hit
mainline, yet nobody appears to have complained. And as it isn't super
easy to implement, just admit defeat and clean up the code.
If somebody wanted to resurrect the feature, the emu8k driver could
serve as a template, but the code would be quite different. But
arguably, this should be done in user space in the first place, as this
doesn't represent a hardware feature (somewhat ironically, the actual
GUS driver has no synth support, and therefore no GUS patch loader).
Note that instead of properly rejecting affected samples, we continue to
just pretend that the feature wasn't requested. This is extremely
questionable behavior, but avoids that possibly unused instruments
suddenly prevent loading the entire file, which would break backwards
compatibility. But at least we log a warning now.
ALSA: emux: fix init of patch_info.truesize in load_data()
The field is explicitly documented to be initialized by the driver
(which it actually is). Also, using patch_info.size would be actually
wrong for 16-bit data, as one field counts samples, while the other
counts bytes.
Merge tag 'asoc-fix-v6.9-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path
In cases where the sof driver is unable to find the firmware and/or
topology file [1], it exits without releasing the i915 runtime
pm wakeref [2]. This results in dmesg warnings[3] during
suspend/resume or driver unbind. Add remove_late() to the failure path
of sof_init_environment so that i915 wakeref is released appropriately
[1]
[ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found.
[ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles
[ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested):
[ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri
[ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg
[ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed.
[ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from:
[ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/
[ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2
Before ACP firmware loading, DSP interrupts are not expected.
Sometimes after reboot, it's observed that before ACP firmware is loaded
false DSP interrupt is reported.
Registering the interrupt handler before acp initialization causing false
interrupts sometimes on reboot as ACP reset is not applied.
Correct the sequence by invoking acp initialization sequence prior to
registering interrupt handler.
Zhang Yi [Tue, 2 Apr 2024 06:20:42 +0000 (14:20 +0800)]
ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume
We got a headphone detection issue after suspend and resume.
And we fixed it by modifying the configuration at es8326_suspend
and invoke es8326_irq at es8326_resume.
We got an error report about headphone type detection and button detection.
We fixed the headphone type detection error by adjusting the debounce timer
configuration. And we fixed the button detection error by disabling the
button detection feature when the headphone are unplugged and enabling it
when headphone are plugged in.
For shutting up spurious KMSAN uninit-value warnings, just replace
kmalloc() calls with kzalloc() for the buffers used for
communications. There should be no real issue with the original code,
but it's still better to cover.
Luke D. Jones [Tue, 2 Apr 2024 01:51:26 +0000 (14:51 +1300)]
ALSA: hda/realtek: cs35l41: Support ASUS ROG G634JYR
Fixes the realtek quirk to initialise the Cirrus amp correctly and adds
related quirk for missing DSD properties. This model laptop has slightly
updated internals compared to the previous version with Realtek Codec
ID of 0x1caf.
Signed-off-by: Luke D. Jones <luke@ljones.dev> Cc: <stable@vger.kernel.org>
Message-ID: <20240402015126.21115-1-luke@ljones.dev> Signed-off-by: Takashi Iwai <tiwai@suse.de>
I Gede Agastya Darma Laksana [Mon, 1 Apr 2024 17:46:02 +0000 (00:46 +0700)]
ALSA: hda/realtek: Update Panasonic CF-SZ6 quirk to support headset with microphone
This patch addresses an issue with the Panasonic CF-SZ6's existing quirk,
specifically its headset microphone functionality. Previously, the quirk
used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design
of a single 3.5mm jack for both mic and audio output effectively. The
device uses pin 0x19 for the headset mic without jack detection.
Following verification on the CF-SZ6 and discussions with the original
patch author, i determined that the update to
ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change
is custom-designed for the CF-SZ6's unique hardware setup, which includes
a single 3.5mm jack for both mic and audio output, connecting the headset
microphone to pin 0x19 without the use of jack detection.
Fixes: 0fca97a29b83 ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk") Signed-off-by: I Gede Agastya Darma Laksana <gedeagas22@gmail.com> Cc: <stable@vger.kernel.org>
Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes the sound not working from internal speakers on
Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID
have been added to cs35l41_hda_property.c and patch_realtek.c.
Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching"
As already anticipated in the original commit, playback was broken for
very short samples. I just didn't expect it to be an actual problem,
because we're talking about less than 1.5 milliseconds here. But clearly
such wavetable samples do actually exist.
The problem was that for such short samples we'd set the current
position beyond the end of the loop, so we'd run off the end of the
sample and play garbage.
This is a bigger (more audible) problem than the original one, which was
that we'd start playback with garbage (whatever was still in the cache),
which would be mostly masked by the note's attack phase.
So revert to the old behavior for now. We'll subsequently fix it
properly with a bigger patch series.
Note that this isn't a full revert - the dead code is not re-introduced,
because that would be silly.
Uwe Kleine-König [Fri, 29 Mar 2024 21:54:42 +0000 (22:54 +0100)]
OSS: dmasound/paula: Mark driver struct with __refdata to prevent section mismatch
As described in the added code comment, a reference to .exit.text is ok
for drivers registered via module_platform_driver_probe(). Make this
explicit to prevent the following section mismatch warning
Stephen Lee [Tue, 26 Mar 2024 01:01:31 +0000 (18:01 -0700)]
ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw
In snd_soc_info_volsw(), mask is generated by figuring out the index of
the most significant bit set in max and converting the index to a
bitmask through bit shift 1. Unintended wraparound occurs when max is an
integer value with msb bit set. Since the bit shift value 1 is treated
as an integer type, the left shift operation will wraparound and set
mask to 0 instead of all 1's. In order to fix this, we type cast 1 as
`1ULL` to prevent the wraparound.
Pierre-Louis Bossart [Mon, 25 Mar 2024 22:18:17 +0000 (17:18 -0500)]
ASoC: rt-sdw*: add __func__ to all error logs
The drivers for Realtek SoundWire codecs use similar logs, which is
problematic to analyze problems reported by CI tools, e.g. "Failed to
get private value: 752001 => 0000 ret=-5". It's not uncommon to have
several Realtek devices on the same platform, having the same log
thrown makes support difficult.
This patch adds __func__ to all error logs which didn't already
include it.
No functionality change, only error logs are modified.
Simon Trimmer [Thu, 28 Mar 2024 12:13:55 +0000 (12:13 +0000)]
ALSA: hda: cs35l56: Add ACPI device match tables
Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules
to be loaded on boot so that Serial Multi Instantiate can add the
devices to the bus and begin the driver init sequence.
Christoffer Sandberg [Thu, 28 Mar 2024 10:27:57 +0000 (11:27 +0100)]
ALSA: hda/realtek - Fix inactive headset mic jack
This patch adds the existing fixup to certain TF platforms implementing
the ALC274 codec with a headset jack. It fixes/activates the inactive
microphone of the headset.
Gergo Koteles [Tue, 26 Mar 2024 16:18:48 +0000 (17:18 +0100)]
ALSA: hda/tas2781: remove useless dev_dbg from playback_hook
The debug message "Playback action not supported: action" is not useful,
because the action was previously printed, and the list of supported
actions are intentional.
Remove the debug statement from the default switch case.
Gergo Koteles [Tue, 26 Mar 2024 16:18:45 +0000 (17:18 +0100)]
ALSA: hda/tas2781: remove digital gain kcontrol
The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A)
register. Unfortunately the tas2563 does not have DVC_LVL, but has
INT_MASK0 in 0x1A, which has been misused so far.
Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer
volume control") the volume of the tas2781 amplifiers can be controlled
by the master volume, so this digital gain kcontrol is not needed.
Arnd Bergmann [Tue, 26 Mar 2024 22:38:07 +0000 (23:38 +0100)]
ALSA: aoa: avoid false-positive format truncation warning
clang warns about what it interprets as a truncated snprintf:
sound/aoa/soundbus/i2sbus/core.c:171:6: error: 'snprintf' will always be truncated; specified size is 6, but format string expands to at least 7 [-Werror,-Wformat-truncation-non-kprintf]
The actual problem here is that it does not understand the special
%pOFn format string and assumes that it is a pointer followed by
the string "OFn", which would indeed not fit.
Slightly increasing the size of the buffer to its natural alignment
avoids the warning, as it is now long enough for the correct and
the incorrect interprations.
Fixes: b917d58dcfaa ("ALSA: aoa: Convert to using %pOFn instead of device_node.name") Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Message-ID: <20240326223825.4084412-9-arnd@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Tue, 26 Mar 2024 13:03:19 +0000 (13:03 +0000)]
ASoC: SOF: ipc4/Intel: Fix delay reporting
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The current version of delay reporting code can report incorrect
values when paired with a firmware which enables this feature.
Unfortunately there are several smaller issues that needed to be addressed
to correct the behavior:
Wrong information was used for the host side of counter
For MTL/LNL used incorrect (in a sense that it was verified only on MTL)
link side counter function.
The link side counter needs compensation logic if pause/resume is used.
The offset values were not refreshed from firmware.
Finally, not strictly connected, but the ALSA buffer size needs to be
constrained to avoid constant xrun from media players (like mpv)
The series applies cleanly for 6.9 and 6.8.y stable, but older stable
would need manual backport, but it is questionable if it is needed as
MTL/LNL is missing features.
Duoming Zhou [Tue, 26 Mar 2024 09:42:38 +0000 (17:42 +0800)]
ALSA: sh: aica: reorder cleanup operations to avoid UAF bugs
The dreamcastcard->timer could schedule the spu_dma_work and the
spu_dma_work could also arm the dreamcastcard->timer.
When the snd_pcm_substream is closing, the aica_channel will be
deallocated. But it could still be dereferenced in the worker
thread. The reason is that del_timer() will return directly
regardless of whether the timer handler is running or not and
the worker could be rescheduled in the timer handler. As a result,
the UAF bug will happen. The racy situation is shown below:
In order to mitigate this bug and other possible corner cases,
call mod_timer() conditionally in run_spu_dma(), then implement
PCM sync_stop op to cancel both the timer and worker. The sync_stop
op will be called from PCM core appropriately when needed.
Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Duoming Zhou <duoming@zju.edu.cn>
Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the `DEFINE_FLEX()` helper for an on-stack definition of a
flexible structure where the size of the flexible-array member
is known at compile-time, and refactor the rest of the code,
accordingly.
So, with these changes, fix the following warning:
sound/firewire/amdtp-stream.c:1184:46: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end]
Link: https://github.com/KSPP/linux/issues/202 Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Message-ID: <ZgIsBqoMb7p3fMDr@neat> Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original timestamp is built base on windows epoch time which is not
fit for Linux system and difficult to be used for kernel debugging. This
patch adopts syslog timestamp so that we can simply use dmesg to check
the timestamp between fw and kernel.
Signed-off-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://msgid.link/r/20240322112703.4549-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Simon Trimmer [Mon, 25 Mar 2024 14:44:50 +0000 (14:44 +0000)]
ASoC: cs-amp-lib: Check for no firmware controls when writing calibration
When a wmfw file has not been loaded the firmware control descriptions
necessary to write a stored calibration are not present. In this case
print a more descriptive error message.
The message is logged at info level because it is not fatal, and does
not necessarily imply that anything is broken.
Peter Ujfalusi [Fri, 22 Mar 2024 11:25:04 +0000 (13:25 +0200)]
ASoC: SOF: Intel: hda-dsp: Skip IMR boot on ACE platforms in case of S3 suspend
SoCs with ACE architecture are tailored to use s2idle instead deep (S3)
suspend state and the IMR content is lost when the system is forced to
enter even to S3.
When waking up from S3 state the IMR boot will fail as the content is lost.
Set the skip_imr_boot flag to make sure that we don't try IMR in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://msgid.link/r/20240322112504.4192-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:13 +0000 (15:08 +0200)]
ALSA: hda: Add pplcllpl/u members to hdac_ext_stream
The pplcllpl/u can be used to save the Link Connection Linear Link
Position register value to be used for compensation of the LLP register
value in case the counter is not reset (after pause/resume or
stop/start without closing the stream).
The LLP can be used along with PPHCLDP to calculate delay caused by the DSP
processing for HDA links.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-17-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:12 +0000 (15:08 +0200)]
ASoC: SOF: ipc4-pcm: Correct the delay calculation
This patch improves the delay calculation by relying on the
LLP (Linear Link Position) on the DAI side and the
LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA
position as LPIB, but with a linear count instead of a position in the
ALSA ring buffer. The LDP values are provided in bytes and must be
converted to frames. The difference in units means that the host counter
will wrap earlier than the LLP. We need to wrap the LLP at the same
boundary as the host counter.
The ASoC framework relies on separate pointer and delay callback.
Measurement errors can be reduced by processing all the counter values in
the pointer callback. The delay value is stored, and will be reported to
higher levels in the delay callback.
For playback, the firmware provides a stream_start offset to handle
mixing/pause usages, where the DAI might have started earlier than the
PCM device. The delay calculation must be special-cased when the link
counter has not reached the start offset value, i.e. no valid audio has
left the DSP.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:11 +0000 (15:08 +0200)]
ASoC: SOF: sof-pcm: Add pointer callback to sof_ipc_pcm_ops
The IPC specific pointer callback can be used when additional or custom
handling is needed during the pointer calculation, like executing a delay
calculation at the same time to minimize drift between the reported pointer
and the calculated delay.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:10 +0000 (15:08 +0200)]
ASoC: SOF: ipc4-pcm: Invalidate the stream_start_offset in PAUSED state
When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the
stream will be restarted (resume or start) in which case we need to update
the offset from the firmware.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:04 +0000 (15:08 +0200)]
ASoC: SOF: Intel: Set the dai/host get frame/byte counter callbacks
Add implementation for reading the LDP (Linear DMA Position) to be used as
get_host_byte_counter().
The LDP is counting the number of bytes moved between the DSP and host
memory.
Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting
the frames on the link side of the DSP.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:03 +0000 (15:08 +0200)]
ASoC: SOF: Introduce a new callback pair to be used for PCM delay reporting
For delay calculation we need two information:
Number of bytes transferred between the DSP and host memory (ALSA buffer)
Number of frames transferred between the DSP and external device
(link/codec/DMIC/etc).
The reason for the different units (bytes vs frames) on host and dai side
is that the format on the dai side is decided by the firmware and might
not be the same as on the host side, thus the expectation is that the
counter reflects the number of frames.
The kernel know the host side format and in there we have access to the
DMA position which is in bytes.
In a simplified way, the DSP caused delay is the difference between the
two counters.
The existing get_stream_position callback is defined to retrieve the frame
counter on the DAI side but it's name is too generic to be intuitive and
makes it hard to define a callback for the host side.
This patch introduces a new set of callbacks to replace the
get_stream_position and define the host side equivalent:
get_dai_frame_counter
get_host_byte_counter
Subsequent patches will remove the old callback.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:02 +0000 (15:08 +0200)]
ASoC: SOF: Intel: mtl/lnl: Use the generic get_stream_position callback
Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related
defines since it can only work on platforms which have 19 streams because
of the use of 0x948 as base offset for the LLP registers.
The generic hda_dsp_get_stream_hda_link_position() takes the number of
streams into consideration when reading the LLP registers for the stream
and can handle different HDA configurations.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:01 +0000 (15:08 +0200)]
ASoC: SOF: Intel: hda: Implement get_stream_position (Linear Link Position)
When the Linear Link Position is not available in firmware SRAM window we
use the host accessible position registers to read it.
The address of the PPLCLLPL/U registers depend on the number of streams
(playback+capture).
At probe time the pplc_addr is calculated for each stream and we can use
it to read the LLP without the need of address re-calculation.
Set the get_stream_position callback in sof_hda_common_ops for all
platforms:
The callback is used for IPC4 delay calculations only but the register is
a generic HDA register, not tied to any specific IPC version.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Peter Ujfalusi [Thu, 21 Mar 2024 13:08:00 +0000 (15:08 +0200)]
ASoC: SOF: Intel: hda-pcm: Use dsp_max_burst_size_in_ms to place constraint
If the PCM have the dsp_max_burst_size_in_ms set then place a constraint
to limit the minimum buffer time to avoid xruns caused by DMA bursts
spinning on the ALSA buffer.
Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Simon Trimmer [Mon, 25 Mar 2024 14:29:37 +0000 (14:29 +0000)]
ALSA: hda: cs35l56: Raise device name message log level
The system and amplifier names influence which firmware and tuning files
are downloaded to the device; log these values to aid end-user system
support.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240325142937.257869-1-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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