- 13 Dec, 2018 1 commit
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Takashi Iwai authored
Pull refactoring / fixes of HD-audio PM and display power management Signed-off-by:Takashi Iwai <tiwai@suse.de>
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- 11 Dec, 2018 15 commits
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Takashi Sakamoto authored
Fireface 800 is a flagship model of RME GmbH for audio and music units on IEEE 1394 bus, shipped 2004. This model consists of four chips: - TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus - TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus - Xilinx Spartan-3 FPGA XC3S400 - Xilinx High-Performance CPLD XC9572XL This commit adds support Fireface 800. In this time, the support is restricted to its MIDI functionality, thus this commit adds some condition statements to avoid touching streaming functionality. Unlike Fireface 400, Fireface 800 has no functionality to suppress asynchronous transactions for MIDI messages except for unregister of listen address in controller side, thus the feature is available as is. Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by:
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Takashi Sakamoto authored
Content of asynchronous transaction for MIDI messages differs between Fireface 400 and 800. This commit adds a model-specific handler for the transaction and adds arrangement. Signed-off-by:
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Takashi Sakamoto authored
Fireface 400 and 800 have the same mechanism to decide address to which asynchronous transactions are sent for MIDI messages, however they use different registers for controllers to notify higher 4 byte of the address. This commit adds a model-specific parameter to represent the address. Additionally, it corrects some comments. I note that these two models have a difference to enable/disable the transaction. Signed-off-by:
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Takashi Sakamoto authored
As long as investigating packet dumps from Fireface 400/800, a register to receive asynchronous transactions for MIDI messages is the same. For Fireface 800, minor register is used. This commit declares macros for the transactions and obsoletes model-specific parameters. Signed-off-by:
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Takashi Sakamoto authored
Unlike Fireface 400, Fireface 800 have two pair of optical interface for ADAT signal and S/PDIF signal. ADAT signals for the interface are handled for sampling clock source separately. This commit modifies a parser for clock configuration to distinguish these two ADAT signals. Signed-off-by:
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Takashi Sakamoto authored
As long as investigating packet dumps from Fireface 400/800, bits on status registers for clock synchronization are the same. This commit moves a parser for a register of clock configuration to obsolete model-specific operations. Signed-off-by:
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Takashi Sakamoto authored
As long as investigating packet dumps from Fireface 400/800, bits on status registers for clock synchronization are the same. This commit moves a parser for the registers to obsolete model-specific operations. Signed-off-by:
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Takashi Sakamoto authored
As long as investigating packet dumps from Fireface 400/800, status registers for clock synchronization is common. This commit moves some macros for them to header file. Signed-off-by:
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Arnd Bergmann authored
When building without CONFIG_PCI, we can (depending on the architecture) get a link failure: ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined! Adding a compile-time check for PCI gets it to work correctly on 32-bit ARM. Fixes: d99501b8 ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()") Signed-off-by:
Arnd Bergmann <arnd@arndb.de> Signed-off-by:
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Takashi Iwai authored
We've excluded the display_power_control flag for Intel HSW and BDW codecs as the HD-audio controllers of the corresponding platforms take care of the display power as well. But the recent refactoring separates the controller and the codec power accounting, so it's fine to call the display PM even for HSW/BDW codecs. This is less confusing since we can avoid this well-hidden condition. Signed-off-by: -
Takashi Iwai authored
The display power is in unbalance at removing the driver since it misses the snd_hdac_display_power(OFF) call. Acked-by:
Mark Brown <broonie@kernel.org> Signed-off-by:
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Takashi Iwai authored
After the recent refactoring, snd_hdac_display_power() doesn't return any error, hence it can be defined to return void. This makes many error checks redundant and allows us to reduce them gracefully. Signed-off-by: -
Takashi Iwai authored
When an error occurs in azx_probe_continue(), we should release the display power. However, the current code ignores it and releases the display power only for HSW/BDW cases. Fix it. Signed-off-by: -
Takashi Iwai authored
snd_hdac_display_power() can be called even for a HDA controller without DRM binding. The same is true for other helpers, snd_hdac_i915_set_bclk() and snd_hdac_set_codec_wakeup(). So all superfluous AZX_DCAPS_I915_POWERWELL checks in hda_intel.c can be dropped, and the definition of AZX_DCAPS_I915_POWERWELL itself can be removed as well. This simplifies the code a lot. Signed-off-by: -
Takashi Iwai authored
The current HD-audio code manages the DRM audio power via too complex redirections, and this seems even still unbalanced in a corner case as Intel DRM CI has been intermittently reporting. This patch is a big surgery for addressing the complexity and the possible unbalance. Basically the patch changes the display PM in the following ways: - Both HD-audio controller and codec drivers call a single helper, snd_hdac_display_power(). (Formerly, the display power control from a codec was done indirectly via link_power bus ops.) - snd_hdac_display_power() receives the codec address index. For turning on/off from the controller, pass HDA_CODEC_IDX_CONTROLLER. - snd_hdac_display_power() doesn't manage refcounts any longer, but keeps the power status in bitmap. If any of controller or codecs is turned on, the function updates the DRM power state via get_power() or put_power(). Also this refactor allows us more cleanup: - The link_power bus ops is dropped, so there is no longer indirect management, as mentioned in the above. - hdac_device link_power_control flag is moved to hda_codec display_power_control flag, as it's only for HDA legacy. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=106525Signed-off-by:
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- 10 Dec, 2018 4 commits
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Takashi Iwai authored
Back-merge for resolving the conflict of fixup entries added in both branches. Signed-off-by: -
Jian-Hong Pan authored
The ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC and output through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied. Signed-off-by:
Daniel Drake <drake@endlessm.com> Signed-off-by:
Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Jian-Hong Pan authored
The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied. Signed-off-by:
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Chris Chiu authored
The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack sensing and enable use of the internal microphone on this laptop X542UN. However, it's ALC294 so create a new fixup named ALC294_FIXUP_ASUS_MIC to avoid confusion. Signed-off-by:
Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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- 09 Dec, 2018 3 commits
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Takashi Iwai authored
Make unified suspend / resume helpers and call them from both the runtime- and the system-PM callbacks for simplifying code. There are slight changes of call orders, but there shouldn't be any functional difference after refactoring. Signed-off-by: -
Takashi Sakamoto authored
In an initial commit, 'SYNC_STATUS' register is referred to get clock configuration, however this is wrong, according to my local note at hand for reverse-engineering about packet dump. It should be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config() is correctly programmed. This commit fixes the bug. Cc: <stable@vger.kernel.org> # v4.12+ Fixes: 76fdb3a9 ('ALSA: fireface: add support for Fireface 400') Signed-off-by:
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Hui Wang authored
Users reported a mute LED regression on Lenovo X1 Carbon, the root cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE to this machine, then the machine can't apply the fixup of ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup together. Fixes: c4cfcf6f ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Cc: <stable@vger.kernel.org> Signed-off-by:
Hui Wang <hui.wang@canonical.com> Signed-off-by:
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- 07 Dec, 2018 5 commits
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Kailang Yang authored
This patch will enable headset button for new Chrome platform. Signed-off-by:
Kailang Yang <kailang@realtek.com> Signed-off-by:
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Takashi Iwai authored
Extend some structs to add the support for jack button changes. Now snd_hda_jack_add_kctl() receives two more arguments: the jack type and the jack keymaps. Both are optional, and when zero are passed, the function behaves just like before. For reporting button state changes, you'd need to update jack->button_state bits accordingly, typically in the jack callback. Then the value OR'ed with button_state and the jack plug state is passed to snd_jack_report(). Note that currently the code assumes only the one-shot button events, i.e. it tries to send the button release soon after sending the button event. If a driver really supports the button release handling by itself, we may need to introduce some flag to control this behavior in future. Signed-off-by: -
Takashi Iwai authored
For allowing the callee to evaluate the associated jack information and the unsolicited event data, add the new fields to hda_jack_callback. They can be used, for example, to retrieve the headset button state in the callback. Signed-off-by: -
Takashi Iwai authored
Back-merge for applying the more HD-audio quirks on top of the latest code. Signed-off-by: -
Kailang Yang authored
If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. Signed-off-by:
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- 06 Dec, 2018 2 commits
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Rob Herring authored
Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. A couple of open coded iterating thru the child node names are converted to use for_each_child_of_node() instead. Signed-off-by:
Rob Herring <robh@kernel.org> Signed-off-by:
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Rob Herring authored
Convert soundbus uevent and sysfs OF node name and device type usage to use printf specifier and helper functions instead of directly accessing the name and type pointers. This will allow the eventual removal of the pointers. Signed-off-by:
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- 05 Dec, 2018 4 commits
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Chris Chiu authored
Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues with the input from external microphone. The issue can be fixed by the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G. Signed-off-by:
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Chris Chiu authored
Acer AIO Veriton Z4660G with ALC286 codec has issue with the input from external microphones connecting via 'Front Mic' jack. The fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of the headset and fix the audio input issue of external microphone. Signed-off-by:
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Chris Chiu authored
The Acer AIO Aspire C24-860 with ALC286 can't detect the headset microphone. Just like another Acer AIO U27-880, it needs a different pin value for 0x18 and the headset fixup to make headset mic work. Signed-off-by:
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Chris Chiu authored
Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic and internal mic not working either. It needs the similar quirk like Sony laptops to fix headphone jack sensing and enables use of the internal microphone. Unfortunately jack sensing for the headset mic is still not working. Signed-off-by:
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- 03 Dec, 2018 4 commits
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Thierry Reding authored
Tegra186 and Tegra194 contain the same codecs as earlier chips and can be supported using the same patch function. Signed-off-by:
Thierry Reding <treding@nvidia.com> Signed-off-by:
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Thierry Reding authored
Recent devices support more than the 4 codecs that the AZX core will probe by default. Probe up to 8 codecs to make sure all of them are enumerated. Suggested-by:
Sameer Pujar <spujar@nvidia.com> Signed-off-by:
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Hui Peng authored
If a USB sound card reports 0 interfaces, an error condition is triggered and the function usb_audio_probe errors out. In the error path, there was a use-after-free vulnerability where the memory object of the card was first freed, followed by a decrement of the number of active chips. Moving the decrement above the atomic_dec fixes the UAF. [ The original problem was introduced in 3.1 kernel, while it was developed in a different form. The Fixes tag below indicates the original commit but it doesn't mean that the patch is applicable cleanly. -- tiwai ] Fixes: 362e4e49 ("ALSA: usb-audio - clear chip->probing on error exit") Reported-by:
Hui Peng <benquike@gmail.com> Reported-by:
Mathias Payer <mathias.payer@nebelwelt.net> Signed-off-by:
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Takashi Iwai authored
We've got a regression report for some Thinkpad models (at least T570s) which shows the too low speaker output volume. The bisection leaded to the commit 61fcf8ec ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform"), and it's basically adding the two pin configurations for the dock, and looks harmless. The real culprit seems, though, that the DAC assignment for the speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14 to be coupled with DAC NID 0x03. When more pins are configured by the commit above, the auto-parser changes the DAC assignment, and this resulted in the regression. As a workaround, just provide the fixed pin / DAC mapping table for this Thinkpad fixup function. It's no generic solution, but the problem itself is pretty much device-specific, so must be good enough. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304 Fixes: 61fcf8ec ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform") Cc: <stable@vger.kernel.org> Reported-and-tested-by:
Jeremy Cline <jcline@redhat.com> Signed-off-by:
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- 29 Nov, 2018 2 commits
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Takashi Iwai authored
This is a series of patches for conversion to LEDs audio-mute trigger. It's based on 4.20-rc3 to be an immutable branch. Signed-off-by: -
Takashi Iwai authored
As addressed in alsa-lib (commit b420056604f0), we need to fix the case where the evaluation of PCM interval "(x x+1]" leading to -EINVAL. After applying rules, such an interval may be translated as "(x x+1)". Fixes: ff2d6acd ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max") Cc: <stable@vger.kernel.org> Signed-off-by:
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