November 2021 - Alsa-devel - mailman.alsa-project.org

hw: add option to specify dynamic DAPM routing to apply when the plugin is opened
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-lib pull request #192 was opened from bku-sue: The kernel's sound soc subsystem has a DAPM functionality, which makes complex audio routing easy, and most of it is done inside the kernel. However, in the case of dynamic routing (when there is a multiplexer in the signal path), the multiplexer is exposed as an enum MIXER control which must be set from userspace. As the selection of these controls directly influences what audio path is captured/played, the sensible thing to do is to create multiple HW plugins in the ALSA conf, each referring to the same HW device number, but with different routing parameters. Request URL : https://github.com/alsa-project/alsa-lib/pull/192 Patch URL : https://github.com/alsa-project/alsa-lib/pull/192.patch Repository URL: https://github.com/alsa-project/alsa-lib

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hw: add "min_rate" and "max_rate" as alternatives to single "rate" parameter
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-lib pull request #191 was opened from bku-sue: Our typical use case is to limit the rate to between 44100 and 48000, as with these boundaries the resampling becomes easier, than if only allowing 48000. Request URL : https://github.com/alsa-project/alsa-lib/pull/191 Patch URL : https://github.com/alsa-project/alsa-lib/pull/191.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Token to create an account on the wiki
by Lucas 15 Nov '21

15 Nov '21

I was hoping to add some Edirol/Roland devices to the wiki. Signing up at https://alsa-project.org/main/index.php?title=Special:UserLogin&type=signup… says ask the mailing list for a token. Thanks, Lucas

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[PATCH] ALSA: hda: intel: More comprehensive PM runtime setup for controller driver
by Takashi Iwai 15 Nov '21

15 Nov '21

Currently we haven't explicitly enable and allow/forbid the runtime PM at the probe and the remove phases of HD-audio controller driver, and this was the reason of a GPF mentioned in the commit e81478bbe7a1 ("ALSA: hda: fix general protection fault in azx_runtime_idle"); namely, even after the resources are released, the runtime PM might be still invoked by the bound graphics driver during the remove of the controller driver. Although we've fixed it by clearing the drvdata reference, it'd be also better to cover the runtime PM issue more properly. This patch adds a few more pm_runtime_*() calls at the probe and the remove time for setting and cleaning up the runtime PM. Particularly, now more explicitly pm_runtime_enable() and _disable() get called as well as pm_runtime_forbid() call at the remove callback, so that a use-after-free should be avoided. Reported-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Tested-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211110210307.1172004-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe51163f2d82..45e85180048c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1347,8 +1347,14 @@ static void azx_free(struct azx *chip) if (hda->freed) return; - if (azx_has_pm_runtime(chip) && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) { pm_runtime_get_noresume(&pci->dev); + pm_runtime_disable(&pci->dev); + pm_runtime_set_suspended(&pci->dev); + pm_runtime_forbid(&pci->dev); + pm_runtime_dont_use_autosuspend(&pci->dev); + } + chip->running = 0; azx_del_card_list(chip); @@ -2322,6 +2328,8 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); + pm_runtime_set_active(&pci->dev); + pm_runtime_enable(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.31.1

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[PATCH] ALSA: hda: fix general protection fault in azx_runtime_idle
by Kai Vehmanen 15 Nov '21

15 Nov '21

Fix a corner case between PCI device driver remove callback and runtime PM idle callback. Following sequence of events can happen: - at azx_create, context is allocated with devm_kzalloc() and stored as pci_set_drvdata() - user-space requests to unbind audio driver - dd.c:__device_release_driver() calls PCI remove - pci-driver.c:pci_device_remove() calls the audio driver azx_remove() callback and this is completed - pci-driver.c:pm_runtime_put_sync() leads to a call to rpm_idle() which again calls azx_runtime_idle() - the azx context object, as returned by dev_get_drvdata(), is no longer valid -> access fault in azx_runtime_idle when executing struct snd_card *card = dev_get_drvdata(dev); chip = card->private_data; if (chip->disabled || hda->init_failed) This was discovered by i915_module_load test with 5.15.0 based linux-next tree. Example log caught by i915_module_load test with linux-next https://intel-gfx-ci.01.org/tree/linux-next/ <4> [264.038232] general protection fault, probably for non-canonical address 0x6b6b6b6b6b6b73f0: 0000 [#1] PREEMPT SMP NOPTI <4> [264.038248] CPU: 0 PID: 5374 Comm: i915_module_loa Not tainted 5.15.0-next-20211109-gc8109c2ba35e-next-20211109 #1 [...] <4> [264.038267] RIP: 0010:azx_runtime_idle+0x12/0x60 [snd_hda_intel] [...] <4> [264.038355] Call Trace: <4> [264.038359] <TASK> <4> [264.038362] __rpm_callback+0x3d/0x110 <4> [264.038371] rpm_idle+0x27f/0x380 <4> [264.038376] __pm_runtime_idle+0x3b/0x100 <4> [264.038382] pci_device_remove+0x6d/0xa0 <4> [264.038388] device_release_driver_internal+0xef/0x1e0 <4> [264.038395] unbind_store+0xeb/0x120 <4> [264.038400] kernfs_fop_write_iter+0x11a/0x1c0 Fix the issue by setting drvdata to NULL at end of azx_remove(). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) Some non-persistent direct links showing the bug trigger on different platforms with linux-next 20211109: - https://intel-gfx-ci.01.org/tree/linux-next/next-20211109/fi-tgl-1115g4/igt… - https://intel-gfx-ci.01.org/tree/linux-next/next-20211109/fi-jsl-1/igt@i915… Notably with 20211110 linux-next, the bug does not trigger: - https://intel-gfx-ci.01.org/tree/linux-next/next-20211110/fi-tgl-1115g4/igt… diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7762718cf429..b90c817e3f6f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2364,6 +2364,7 @@ static void azx_remove(struct pci_dev *pci) cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); + pci_set_drvdata(pci, NULL); snd_card_free(card); } } base-commit: 6322ec8d0de924cf9672b23c1b5052afafc2f03b -- 2.33.0

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ucm2: Add UCM support for Dell Desktop
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-ucm-conf pull request #120 was opened from shumingfan: The Dell Desktop uses two USB audio codecs as front and rear panel. The front UAC has two outputs of speaker/headphone and one input of headset mic. The rear UAC has one output of line-out. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/120 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/120.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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HP Probook Mute volume + mic LED indicators not working (Audio codec = ALC236)
by GitHub issues - opened 14 Nov '21

14 Nov '21

alsa-project/alsa-lib issue #190 was opened from antintin0: Tested on kernel 5.13 and 5.15.2 Not sure if this is the right place to ask, but no one else seems to be able to help. I have an HP Probook x360 435 g7 with the Realtek ALC236 codec and my mute volume + mute mic LED indicators aren't working (they do on windows). I found there were some patches mentioning mute LEDs on ALC236 on the below links: https://patchwork.kernel.org/project/alsa-devel/patch/20210701091417.9696-1… https://mailman.alsa-project.org/pipermail/alsa-devel/2021-November/192225.… But obviously they're not working for me, so I tried following this: https://wiki.archlinux.org/title/Laptop#Audio_mute_LED Theres nothing specific for ALC236 but I tried pretty much all the relevant parameters for the snd-hda-intel module option "model" under "ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)" and none worked. Any help would be appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/190 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ALSA: hda/realtek: Add quirk for ASRock NUC Box 1100
by Werner Sembach 14 Nov '21

14 Nov '21

This applies a SND_PCI_QUIRK(...) to the ASRock NUC Box 1100 series. This fixes the issue of the headphone jack not being detected unless warm rebooted from a certain other OS. When booting a certain other OS some coeff settings are changed that enable the audio jack. These settings are preserved on a warm reboot and can be easily dumped. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f1727faec69..701c80ed83dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6521,6 +6521,27 @@ static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *co alc_write_coef_idx(codec, 0x45, 0x5089); } +static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { + WRITE_COEF(0x1a, 0x9003), WRITE_COEF(0x1b, 0x0e2b), WRITE_COEF(0x37, 0xfe06), + WRITE_COEF(0x38, 0x4981), WRITE_COEF(0x45, 0xd489), WRITE_COEF(0x46, 0x0074), + WRITE_COEF(0x49, 0x0149), + {} +}; + +static void alc233_fixup_no_audio_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The audio jack input and output is not detected on the ASRock NUC Box + * 1100 series when cold booting without this fix. Warm rebooting from a + * certain other OS makes the audio functional, as COEF settings are + * preserved in this case. This fix sets these altered COEF values as + * the default. + */ + alc_process_coef_fw(codec, alc233_fixup_no_audio_jack_coefs); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6740,6 +6761,7 @@ enum { ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC233_FIXUP_NO_AUDIO_JACK, }; static const struct hda_fixup alc269_fixups[] = { @@ -8460,6 +8482,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, }, + [ALC233_FIXUP_NO_AUDIO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_no_audio_jack, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8894,6 +8920,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), -- 2.25.1

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Possible issue with Intel 200 PCH HD on X299 AORUS Gaming 7
by Ramon Fried 13 Nov '21

13 Nov '21

Hi all. Apparently on Linux Headphones and Line are switched. Dual booting Windows on the same machine proves that it's indeed connected correctly. I initially thought it was related to Pulseaudio, but together with Pulseaudio guys help we figured it's probably in ALSA. http://alsa-project.org/db/?f=535021978814678ea328b0d3a053ba3cbd39b709 Where should I open a bug ? Is it kernel related, if user-space, to which ALSA component should I open a bug to ? Thanks, Ramon

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[PATCH v3 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 12 Nov '21

12 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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