- 22 Mar, 2021 2 commits
-
-
Hui Wang authored
We found the alc_update_headset_mode() is not called on some machines when unplugging the headset, as a result, the mode of the ALC_HEADSET_MODE_UNPLUGGED can't be set, then the current_headset_type is not cleared, if users plug a differnt type of headset next time, the determine_headset_type() will not be called and the audio jack is set to the headset type of previous time. On the Dell machines which connect the dmic to the PCH, if we open the gnome-sound-setting and unplug the headset, this issue will happen. Those machines disable the auto-mute by ucm and has no internal mic in the input source, so the update_headset_mode() will not be called by cap_sync_hook or automute_hook when unplugging, and because the gnome-sound-setting is opened, the codec will not enter the runtime_suspend state, so the update_headset_mode() will not be called by alc_resume when unplugging. In this case the hp_automute_hook is called when unplugging, so add update_headset_mode() calling to this function. Cc: <stable@vger.kernel.org> Signed-off-by:
Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20210320091542.6748-2-hui.wang@canonical.comSigned-off-by:
Takashi Iwai <tiwai@suse.de>
-
Hui Wang authored
We found a recording issue on a Dell AIO, users plug a headset-mic and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. On this machine, the internal mic is not connected to the codec, the "Input Source" is headset mic by default. And when users plug a headset, the determine_headset_type() will be called immediately, the codec on this AIO is alc274, the delay time for this codec in the determine_headset_type() is only 80ms, the delay is too short to correctly determine the headset type, the fail rate is nearly 99% when users plug the headset with the normal speed. Other codecs set several hundred ms delay time, so here I change the delay time to 850ms for alc2x4 series, after this change, the fail rate is zero unless users plug the headset slowly on purpose. Cc: <stable@vger.kernel.org> Signed-off-by:
-
- 18 Mar, 2021 2 commits
-
-
Colin Ian King authored
The shifting of the u8 integer device by 24 bits to the left will be promoted to a 32 bit signed int and then sign-extended to a 64 bit unsigned long. In the event that the top bit of device is set then all then all the upper 32 bits of the unsigned long will end up as also being set because of the sign-extension. Fix this by casting device to an unsigned long before the shift. Addresses-Coverity: ("Unintended sign extension") Fixes: a07df82c ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk") Signed-off-by:Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.comSigned-off-by:
-
Takashi Iwai authored
Merge tag 'asoc-fix-v5.12-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.12 Quite a lot of mostly platform specific fixes here, the only one which is generic is a fix for regressions on devices with more complex clocking support with simple-card. There's also a few new device IDs and platform quirks.
-
- 16 Mar, 2021 8 commits
-
-
Mark Brown authored
Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar@nvidia.com>: With commit 1e30f642 ("ASoC: simple-card-utils: Fix device module clock") simple-card-utils can control MCLK clock for rate updates or enable/disable. But this is breaking some platforms where it is expected that codec drivers would actually handle the MCLK clock. One such example is following platform. - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts" In above case codec, wm8904, is using internal PLL and configures sysclk based on fixed MCLK input. In such cases it is expected that, required PLL output or sysclk, is just passed via set_sysclk() callback and card driver need not actually update MCLK rate. Instead, codec can take ownership of this clock and do the necessary configuration. So the original commit is reverted and codec driver for rt5659 is updated to fix my board which has this codec. Sameer Pujar (2): ASoC: simple-card-utils: Do not handle device clock ASoC: rt5659: Update MCLK rate in set_sysclk() sound/soc/codecs/rt5659.c | 5 +++++ sound/soc/generic/simple-card-utils.c | 13 +++++++------ 2 files changed, 12 insertions(+), 6 deletions(-) -- 2.7.4
-
Jeremy Szu authored
The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by:
Jeremy Szu <jeremy.szu@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210316094236.89028-1-jeremy.szu@canonical.comSigned-off-by:
-
Shengjiu Wang authored
Add compatible string for new added platforms which support spdif module. They are i.MX8QXP, i.MX8MM, i.MX8MN, i.MX8MQ. Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Link: https://lore.kernel.org/r/1615884053-4264-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by:
Mark Brown <broonie@kernel.org>
-
Bard Liao authored
We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by:
Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by:
Rander Wang <rander.wang@intel.com> Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.comSigned-off-by:
-
Sameer Pujar authored
Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by:
Michael Walle <michael@walle.cc> Signed-off-by:
Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.comSigned-off-by:
-
Sameer Pujar authored
This reverts commit 1e30f642 ("ASoC: simple-card-utils: Fix device module clock"). The original patch ended up breaking following platform, which depends on set_sysclk() to configure internal PLL on wm8904 codec and expects simple-card-utils to not update the MCLK rate. - "arch/arm64/boot/dts/freescale/fsl-ls1028a-kontron-sl28-var3-ads2.dts" It would be best if codec takes care of setting MCLK clock via DAI set_sysclk() callback. Reported-by:
-
Jeremy Szu authored
The HP EliteBook 840 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by:
-
Jeremy Szu authored
The HP EliteBook 840 G8 Notebook PC is using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by:
-
- 15 Mar, 2021 1 commit
-
-
Xiaoliang Yu authored
Built-in microphone and combojack on Xiaomi Notebook Pro (1d72:1701) needs to be fixed, the existing quirk for Dell works well on that machine. Signed-off-by:
Xiaoliang Yu <yxl_22@outlook.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/OS0P286MB02749B9E13920E6899902CD8EE6C9@OS0P286MB0274.JPNP286.PROD.OUTLOOK.COMSigned-off-by:
-
- 14 Mar, 2021 1 commit
-
-
Xiaoliang Yu authored
There is another fix for headset-mic problem on Redmibook (1d72:1602), it also works on Redmibook Air (1d72:1947), which has the same issue. Signed-off-by:
-
- 12 Mar, 2021 3 commits
-
-
Jiaxin Yu authored
This patch correct tdm out bck inverse register to AUDIO_TOP_CON3[3]. Signed-off-by:
Jiaxin Yu <jiaxin.yu@mediatek.com> Link: https://lore.kernel.org/r/1615516005-781-1-git-send-email-jiaxin.yu@mediatek.comSigned-off-by:
-
Takashi Sakamoto authored
When node is removed from IEEE 1394 bus, any transaction fails to the node. In the case, ALSA dice driver doesn't stop isochronous contexts even if they are running. As a result, null pointer dereference occurs in callback from the running context. This commit fixes the bug to release isochronous contexts always. Cc: <stable@vger.kernel.org> # v5.4 or later Fixes: e9f21129 ("ALSA: dice: support AMDTP domain") Signed-off-by:
Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20210312093407.23437-1-o-takashi@sakamocchi.jpSigned-off-by:
-
Hui Wang authored
Recently we found the micmute led init state is not correct after freshly installing the ubuntu linux on a Lenovo AIO machine. The internal mic is not muted, but the micmute led is on and led mode is 'follow mute'. If we mute internal mic, the led is keeping on, then unmute the internal mic, the led is off. And from then on, the micmute led will work correctly. So the micmute led init state is not correct. The led is controlled by codec gpio (ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), in the patch_realtek, the gpio data is set to 0x4 initially and the led is on with this data. In the hda_generic, the led_value is set to 0 initially, suppose users set the 'capture switch' to on from user space and the micmute led should change to be off with this operation, but the check "if (val == spec->micmute_led.led_value)" in the call_micmute_led_update() will skip the led setting. To guarantee the led state will be set by the 1st time of changing "Capture Switch", set -1 to the init led_value. Cc: <stable@vger.kernel.org> Signed-off-by:
-
- 11 Mar, 2021 2 commits
-
-
Srinivasa Rao Mandadapu authored
The max boundary check while parsing dai ids makes sound card registration fail after common up dai ids. Fixes: cd3484f7 ("ASoC: qcom: Fix broken support to MI2S TERTIARY and QUATERNARY") Signed-off-by:
Srinivasa Rao Mandadapu <srivasam@codeaurora.org> Link: https://lore.kernel.org/r/20210311154557.24978-1-srivasam@codeaurora.orgSigned-off-by:
-
Meng Li authored
When initialize cadence qspi controller, it is need to set cqspi to the driver_data field of struct device, because it will be used in function cqspi_remove/suspend/resume(). Otherwise, there will be a crash trace as below when invoking these finctions. Fixes: 31fb632b ("spi: Move cadence-quadspi driver to drivers/spi/") Cc: stable@vger.kernel.org Signed-off-by:
Meng Li <Meng.Li@windriver.com> Link: https://lore.kernel.org/r/20210311091220.3615-1-Meng.Li@windriver.comSigned-off-by:
-
- 10 Mar, 2021 21 commits
-
-
Mark Brown authored
Merge series "ASoC: sdm845: array out of bound issues" from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>: During testing John Stultz and Amit reported few array our bound issues after enabling bound sanitizer This patch series attempts to fix those! changes since v1: - make sure the wcd is not de-referenced without intialization Srinivas Kandagatla (3): ASoC: qcom: sdm845: Fix array out of bounds access ASoC: qcom: sdm845: Fix array out of range on rx slim channels ASoC: codecs: wcd934x: add a sanity check in set channel map sound/soc/codecs/wcd934x.c | 6 ++++++ sound/soc/qcom/sdm845.c | 6 +++--- 2 files changed, 9 insertions(+), 3 deletions(-) -- 2.21.0
-
Pan Xiuli authored
The ADSPCS_SPA is Set Power Active bit. To check if DSP is powered down, we need to check ADSPCS_CPA, the Current Power Active bit. Fixes: 747503b1 ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations") Reviewed-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by:
Pan Xiuli <xiuli.pan@linux.intel.com> Signed-off-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210309004127.4940-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
-
Srinivas Kandagatla authored
set channel map can be passed with a channel maps, however if the number of channels that are passed are more than the actual supported channels then we would be accessing array out of bounds. So add a sanity check to validate these numbers! Fixes: a61f3b4f ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Reported-by:
John Stultz <john.stultz@linaro.org> Signed-off-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20210309142129.14182-4-srinivas.kandagatla@linaro.orgSigned-off-by:
-
Srinivas Kandagatla authored
WCD934x has only 13 RX SLIM ports however we are setting it as 16 in set_channel_map, this will lead to array out of bounds error! Orignally caught by enabling USBAN array out of bounds check: Fixes: 5caf64c6 ("ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga") Reported-by:
-
Srinivas Kandagatla authored
Static analysis Coverity had detected a potential array out-of-bounds write issue due to the fact that MAX AFE port Id was set to 16 instead of using AFE_PORT_MAX macro. Fix this by properly using AFE_PORT_MAX macro. Fixes: 1b93a884 ("ASoC: qcom: sdm845: handle soundwire stream") Reported-by:
-
Mark Brown authored
Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal@opensource.cirrus.com>: Hi All, Here is a patch series for reporting to user space jack and button events and add the support for Capture. With some cleanups and fixes along the way. Regards, Lucas Tanure Lucas Tanure (12): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it sound/soc/codecs/cs42l42.c | 437 +++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 41 +++- 2 files changed, 282 insertions(+), 196 deletions(-) -- 2.30.1
-
Peter Robinson authored
In 61fbeb5d the sirf prima/atlas drivers were removed. This cleans up a stray header and some Kconfig entries for the codec that were missed in the process. Fixes: 61fbeb5d (ASoC: remove sirf prima/atlas drivers) Signed-off-by:
Peter Robinson <pbrobinson@gmail.com> Cc: Arnd Bergmann <arnd@arndb.de> Cc: Mark Brown <broonie@kernel.org> Acked-by:
Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20210307162338.1160604-1-pbrobinson@gmail.comSigned-off-by:
-
Jonathan Marek authored
Attempting to use the RX MIX path at 48kHz plays at 96kHz, because these controls are incorrectly toggling the first bit of the register, which is part of the FS_RATE field. Fix the problem by using the same method used by the "WSA RX_MIX EC0_MUX" control, which is to use SND_SOC_NOPM as the register and use an enum in the shift field instead. Fixes: 2c4066e5 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Signed-off-by:
Jonathan Marek <jonathan@marek.ca> Reviewed-by:
-
Jonathan Marek authored
An interface can have multiple decimators enabled, so loop over all active decimators. Otherwise only one channel will be unmuted, and other channels will be zero. This fixes recording from dual DMIC as a single two channel stream. Also remove the now unused "active_decimator" field. Fixes: 908e6b1d ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by:
-
Lucas Tanure authored
This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by:
Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.comSigned-off-by:
-
Lucas Tanure authored
dev_pm_ops already enable/disable the codec if not in use Signed-off-by:
-
Lucas Tanure authored
The minimum value is 0x3f (-63dB), which also is mute Signed-off-by:
-
Lucas Tanure authored
Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by:
-
Lucas Tanure authored
The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by:
-
Jon Hunter authored
Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by:
Jon Hunter <jonathanh@nvidia.com> Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.comSigned-off-by:
-
Pierre-Louis Bossart authored
We only unregister the platform device during the .remove operation, but if the probe fails we will never reach this sequence. Suggested-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20210302003410.1178535-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
-
Mark Brown authored
Hi All, Here is a series of rt5640/rt5651 volume-control fixes which I wrote while working on a bytcr-rt5640 UCM profile patch-series adding hardware-volume control to devices using this UCM profile. The UCM series will also work on older kernels, but it works best on kernels with this series applied, giving e.g. finer grained volume control and support for hardware muting the outputs. Regards, Hans Hans de Goede (5): ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' ASoC: Intel: bytcr_rt5640: Add used AIF to the components string sound/soc/codecs/rt5640.c | 106 +++++++++++++++++++++++--- sound/soc/codecs/rt5640.h | 4 + sound/soc/codecs/rt5651.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 ++- 4 files changed, 111 insertions(+), 14 deletions(-) -- 2.30.1 -
Hans de Goede authored
Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.comSigned-off-by:
-
Benjamin Rood authored
According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald@curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdfSigned-off-by:
Benjamin Rood <benjaminjrood@gmail.com> Reviewed-by:
Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-devSigned-off-by:
-
Hans de Goede authored
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by:
-
Hans de Goede authored
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by:
-
