- 22 May, 2023 2 commits
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Takashi Iwai authored
It's reported that the recording started right after the driver probe doesn't work properly, and it turned out that this is related with the codec auto-suspend. Namely, after the probe phase, the usage count goes zero, and the auto-suspend is programmed, but the codec is kept still active until the auto-suspend expiration. When an application (e.g. alsactl) updates the mixer values at this moment, the values are cached but not actually written. Then, starting arecord thereafter also results in the silence because of the missing unmute. The root cause is the handling of "lazy update" mode; when a mixer value is updated *after* the suspend, it should update only the cache and exits. At the resume, the cached value is written to the device, in turn. The problem is that the current code misinterprets the state of auto-suspend as if it were already suspended. Although we can add the check of the actual device state after pm_runtime_get_if_in_use() for catching the missing state, this won't suffice; the second call of regmap_update_bits_check() will skip writing the register because the cache has been already updated by the first call. So we'd need fixes in two different places. OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with pm_runtime_get_if_active() (with ign_usage_count=true). This change implies that the driver takes the pm refcount if the device is still in ACTIVE state and continues the processing. A small caveat is that this will leave the auto-suspend timer. But, since the timer callback itself checks the device state and aborts gracefully when it's active, this won't be any substantial problem. Long story short: we address the missing register-write problem just by replacing the pm_runtime_*() call in snd_hda_keep_power_up(). Fixes: fc4f000b ("ALSA: hda - Fix unexpected resume through regmap code path") Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.comSuggested-by: Cezary Rojewski <cezary.rojewski@intel.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230518113520.15213-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Adam Stylinski authored
This quirk is necessary for surround and other DSP effects to work with the onboard ca0132 based audio chipset for the EVGA X299 dark mainboard. Signed-off-by: Adam Stylinski <kungfujesus06@gmail.com> Cc: <stable@vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=67071 Link: https://lore.kernel.org/r/ZGopOe19T1QOwizS@eggsbenedict.adamsnetSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 17 May, 2023 3 commits
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Nikhil Mahale authored
These IDs are for AD102, AD103, AD104, AD106, and AD107 gpus with audio functions that are largely similar to the existing ones. Tested audio using gnome-settings, over HDMI, DP-SST and DP-MST connections on AD106 gpu. Signed-off-by: Nikhil Mahale <nmahale@nvidia.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230517090736.15088-1-nmahale@nvidia.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Arnd Bergmann authored
Two functions are defined and used in pcm_oss.c but also optionally used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS is disabled, this causes a warning as the functions are not static and have no prototype: sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes] sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes] Avoid this by making the prototypes unconditional. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.orgSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Arnd Bergmann authored
snd_cs46xx_download_image() was originally called from dsp_spos.c, but is now local to cs46xx_lib.c. Mark it as 'static' to avoid a warning about it lacking a declaration, and '__maybe_unused' to avoid a warning about it being unused when CONFIG_SND_CS46XX_NEW_DSP is disabled: sound/pci/cs46xx/cs46xx_lib.c:534:5: error: no previous prototype for 'snd_cs46xx_download_image' Fixes: 89f157d9 ("[ALSA] cs46xx - Fix PM resume") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-1-arnd@kernel.orgSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 16 May, 2023 2 commits
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Takashi Iwai authored
get_line_out_pfx() may trigger an Oops by overflowing the static array with more than 8 channels. This was reported for MacBookPro 12,1 with Cirrus codec. As a workaround, extend for the 9.1 channels and also fix the potential Oops by unifying the code paths accessing the same array with the proper size check. Reported-by: Olliver Schinagl <oliver@schinagl.nl> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/64d95eb0-dbdb-cff8-a8b1-988dc22b24cd@schinagl.nl Link: https://lore.kernel.org/r/20230516184412.24078-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Merge tag 'asoc-fix-v6.4-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v6.4 More fixes that came in since the merge window, the bulk of which are for the SOF code, I suspect as a result of the wide usage, active development and large code size rather than huge quality problems. There's also a couple of MAINTAINERS updates and some new device quirks.
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- 15 May, 2023 20 commits
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Mark Brown authored
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: With additional testing with multiple links and multiple DAI types, we found a couple of mistakes with refcounts, base address, missing initialization. A new helper was also added due to a change in the SoundWire programming sequences, with the host driver in charge of setting up the DMA channel mapping instead of the firmware.
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Ranjani Sridharan authored
The memory allocated for the tuples array assumes that there's 1 instance of all tokens already. So for those tokens that have multiple instances in topology, we need to exclude the initial instance that has already been accounted for. Fixes: 4fdef47a ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230515085200.17094-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Ranjani Sridharan authored
Using the same token ID for both input and output format pin index results in collisions and incorrect pin index getting parsed from topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.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: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Paul Olaru <paul.olaru@oss.nxp.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230515104403.32207-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Mark Brown authored
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>: Three patch to correct error path PM runtime handling in few places. Regards, Peter --- Pierre-Louis Bossart (3): ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions ASoC: SOF: pcm: fix pm_runtime imbalance in error handling ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling sound/soc/sof/debug.c | 4 ++-- sound/soc/sof/pcm.c | 11 ++++++----- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.40.1
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Lucas Tanure authored
I'm leaving Cirrus Logic, and will no longer have access to hardware and documentation necessary to be effective in a maintainership role. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com Link: https://lore.kernel.org/r/20230512154503.741718-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org
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Simon Trimmer authored
Flush the SoundWire interrupt handler work instead of cancelling it. When a SoundWire interrupt is triggered the pm_runtime is held until the work has completed. It's therefore unsafe to cancel the work, it must be flushed. Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com Link: https://lore.kernel.org/r/20230512144237.739000-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org
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Ranjani Sridharan authored
Topology could have more instances of the tokens being searched for than the number of sets that need to be copied. Stop copying token after the limit of number of token instances has been reached. This worked before only by chance as we had allocated more size for the tuples array than the number of actual tokens being parsed. Fixes: 7006d20e ("ASoC: SOF: Introduce IPC3 ops") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512114630.24439-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Kai Vehmanen authored
If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey@chromium.org Link: https://github.com/thesofproject/linux/issues/4274 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 Reviewed-by: Curtis Malainey <cujomalainey@chromium.org Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Krzysztof Kozlowski authored
Remove Krzysztof Kozlowski from maintainer of Samsung SoC Audio drivers and change the status to maintenance (no one is reality being paid for looking at this). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org Reviewed-by: Alim Akhtar <alim.akhtar@samsung.com Link: https://lore.kernel.org/r/20230513090228.4340-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org
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Douglas Anderson authored
When devm runs function in the "remove" path for a device it runs them in the reverse order. That means that if you have parts of your driver that aren't using devm or are using "roll your own" devm w/ devm_add_action_or_reset() you need to keep that in mind. The mt8186 audio driver didn't quite get this right. Specifically, in mt8186_init_clock() it called mt8186_audsys_clk_register() and then went on to call a bunch of other devm function. The caller of mt8186_init_clock() used devm_add_action_or_reset() to call mt8186_deinit_clock() but, because of the intervening devm functions, the order was wrong. Specifically at probe time, the order was: 1. mt8186_audsys_clk_register() 2. afe_priv->clk = devm_kcalloc(...) 3. afe_priv->clk[i] = devm_clk_get(...) At remove time, the order (which should have been 3, 2, 1) was: 1. mt8186_audsys_clk_unregister() 3. Free all of afe_priv->clk[i] 2. Free afe_priv->clk The above seemed to be causing a use-after-free. Luckily, it's easy to fix this by simply using devm more correctly. Let's move the devm_add_action_or_reset() to the right place. In addition to fixing the use-after-free, code inspection shows that this fixes a leak (missing call to mt8186_audsys_clk_unregister()) that would have happened if any of the syscon_regmap_lookup_by_phandle() calls in mt8186_init_clock() had failed. Fixes: 55b423d5 ("ASoC: mediatek: mt8186: support audio clock control in platform driver") Signed-off-by: Douglas Anderson <dianders@chromium.org Link: https://lore.kernel.org/r/20230511092437.1.I31cceffc8c45bb1af16eb613e197b3df92cdc19e@changeid Signed-off-by: Mark Brown <broonie@kernel.org
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Peter Ujfalusi authored
The commands in sof_ipc_dai_config.flags are encoded as bits: 1 (bit0) - hw_params 2 (bit1) - hw_free 4 (bit2) - pause These are commands, they cannot be combined as one would assume, for example 3 (bit0 | bit1) is invalid. This can happen right at the second start of a stream as at the end of the first stream we set the hw_free command (bit1) and on the second start we would OR on top of it the hw_params (bit0). Fixes: b66bfc3a ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Link: https://lore.kernel.org/r/20230512110317.5180-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
When a firmware IPC error happens during a pm_runtime suspend, we ignore the error and suspend anyways. However, the code unconditionally increases the runtime_pm counter. This results in a confusing configuration where the code will suspend, resume but never suspend again due to the use of pm_runtime_get_noresume(). The intent of the counter increase was to prevent entry in D3, but if that transition to D3 is already started it cannot be stopped. In addition, there's no point in that case in trying to prevent anything, the firmware error is handled and the next resume will re-initialize the firmware completely. This patch changes the logic to prevent suspend when the device is pm_runtime active and has a use_count > 0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
These registers enable the HDaudio DMA hardware to split/merge data from different PDIs, possibly on different links. This capability exists for all types of HDaudio extended links, but for now is only required for SoundWire. In the SSP/DMIC case, the IP is programmed by the DSP firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
We defined the values but never initialized it for SoundWire/SSP, fix this miss. A Fixes: tag is not provided as instance_offset was not used so far, so nothing was really broken. This patch is only required for the SoundWire support in the following patch. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
We mix the use of hlink->ml_addr and the 'ml_addr' parameter. It's the same thing, let's align on using the parameter. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
The base_ptr value needs to be derived from the remap_addr pointer, not the ml_addr. This base_ptr was used only in debug logs that were so far not contributed upstream so the issue was not detected. It needs to be fixed for SoundWire support on LunarLake. Fixes: 17c9b6ec ("ASoC: SOF: Intel: hda-mlink: add structures to parse ALT links") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
Same functionality as for DMIC/SSP with different ID. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Rander Wang <rander.wang@intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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Pierre-Louis Bossart authored
In hindsight it was a very bad idea to use the same refcount for Extended and 'legacy' HDaudio multi-links. The existing solution only powers-up the first sublink, which causes SoundWire and SSP tests to fail when more than one DAI is used concurrently. Solving this problem requires per-sublink refcounting, as suggested in this patch. The existing refcounting remains for 'legacy' HdAudio links, mainly to avoid changing the obscure programming sequence in snd_hdac_ext_bus_link_put(). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com Link: https://lore.kernel.org/r/20230512174611.84372-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org
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- 12 May, 2023 5 commits
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Kai-Heng Feng authored
There's yet another laptop that needs the fixup to enable mute and micmute LEDs. So do it accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230512083417.157127-1-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Ryan C. Underwood authored
The Pavilion 15 line has B&O top speakers similar to the x360 and applying the same profile produces good sound. Without this, the sound would be tinny and underpowered without either applying model=alc295-hp-x360 or booting another OS first. Signed-off-by: Ryan Underwood <nemesis@icequake.net> Fixes: 563785ed ("ALSA: hda/realtek - Add quirk entry for HP Pavilion 15") Link: https://lore.kernel.org/r/ZF0mpcMz3ezP9KQw@icequake.netSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed 48k sample rate like other Line6 models. This patch adds the corresponding entry to line6_parse_audio_format_rate_quirk(). Reported-by: John Humlick <john@humlick.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230512075858.22813-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Dan Carpenter authored
This code was supposed to return an error code if init_stream() failed, but it instead freed dg00x->rx_stream and returned success. This potentially leads to a use after free. Fixes: 9a08067e ("ALSA: firewire-digi00x: support AMDTP domain") Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org> Link: https://lore.kernel.org/r/c224cbd5-d9e2-4cd4-9bcf-2138eb1d35c6@kili.mountainSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Paweł Anikiel authored
Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset-issue/480229 I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan@semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie@kernel.org
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- 11 May, 2023 4 commits
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Alexandru Sorodoc authored
These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ffSigned-off-by: Alexandru Sorodoc <ealex95@gmail.com> Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Vitaly Rodionov authored
Add support for HP EliteBook 835/845/845W/865 G10 laptops with CS35L41 amplifiers on I2C/SPI bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230510142227.32945-1-vitalyr@opensource.cirrus.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Martin Povišer authored
A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin@cutebit.org Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie@kernel.org
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Aidan MacDonald authored
When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org
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- 09 May, 2023 1 commit
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V sujith kumar Reddy authored
Check substream and runtime variables before assigning. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com Link: https://lore.kernel.org/r/20230508070510.6100-1-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown <broonie@kernel.org
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- 08 May, 2023 3 commits
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Shengjiu Wang authored
There is error message when defer probe happens: fsl-micfil-dai 30ca0000.micfil: Unbalanced pm_runtime_enable! Fix the error handler with pm_runtime_enable and add fsl_micfil_remove() for pm_runtime_disable. Fixes: 47a70e6f ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com Link: https://lore.kernel.org/r/1683540996-6136-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org
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Ai Chao authored
Add a quirk for HP EliteDesk 805 to fixup ALC3867 headset MIC no sound. Signed-off-by: Ai Chao <aichao@kylinos.cn> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230506022653.2074343-1-aichao@kylinos.cnSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Luke D. Jones authored
Add quirk for GU603 with 0x1c62 variant of codec. Signed-off-by: Luke D. Jones <luke@ljones.dev> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230505235824.49607-2-luke@ljones.devSigned-off-by: Takashi Iwai <tiwai@suse.de>
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