September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: soc-component.h: tidyup for soc-component.h
by Kuninori Morimoto 20 Sep '19

20 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, GPL explanation is not needed. And it is included from soc.h only. No need to header-test. This patch tidyup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark this is for for-5.4 include/Kbuild | 1 + include/sound/soc-component.h | 6 ------ 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/include/Kbuild b/include/Kbuild index c38f0d4..c6909b5 100644 --- a/include/Kbuild +++ b/include/Kbuild @@ -1052,6 +1052,7 @@ header-test- += sound/soc-acpi-intel-match.h header-test- += sound/soc-dai.h header-test- += sound/soc-dapm.h header-test- += sound/soc-dpcm.h +header-test- += sound/soc-component.h header-test- += sound/sof/control.h header-test- += sound/sof/dai-intel.h header-test- += sound/sof/dai.h diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..a869199 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,16 +3,10 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -#include <sound/soc.h> - /* * Component probe and remove ordering levels for components with runtime * dependencies. -- 2.7.4

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[alsa-devel] RfC / [PATCH] Jack sense support for Medion E4254
by Jan-Marek Glogowski 20 Sep '19

20 Sep '19

Hi everybody, I recently bought the laptop. It has one jack for attaching a headset. Jack sense is announced by the HDA config but not working. Except for some positioning, and color information, the PIN config seemed to be sensible, and everything I tried to override, didn't result in a working jack. Manually unmuting the jack using hda-analyser resulted in working output, so generally the config seemed to work. Reading the patch-realtek code, if found ALC256_FIXUP_ASUS_HEADSET_MODE, which somehow changed the pin 0x19 to be associated with the 0x21 headset out pin, resulting in this dmesg change: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 And for whatever reason jack sense is now working, at least for some normal output-only headset (currently I don't have one included a mic with me, so actually can't test that mic volume works correctly). I was just wondering, if my pin values are actually correct, because they still define different association, as I understand it, but the ASUS fix even works with these wrong channels /associations 2 and 4. So the attached patch works for me, but I'm still wondering, if this is the correct solution. FWIW, jack sense works fine in Windows. Anything I might be missing here? Jan-Marek ------- >From 0ccc5f07d8737690bd2df6d88a5af0f1f36d0e40 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski <glogow(a)fbihome.de> Date: Sun, 15 Sep 2019 16:57:28 +0200 Subject: [PATCH] ALSA: hda/realtek: PCI quirk for Medion E4254 The laptop has a combined jack to attach headsets on the right. The BIOS encodes them as two different colored jacks at the front, but otherwise it seems to be configured ok. But any adaption of the pins config on its own doesn't fix the jack detection to work in Linux. Still Windows works correct. This is somehow fixed by chaining ALC256_FIXUP_ASUS_HEADSET_MODE, which seems to register the microphone jack as a headset part and also results in fixing jack sensing, visible in dmesg as: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 Signed-off-by: Jan-Marek Glogowski <glogow(a)fbihome.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1ddfd2fac52..448e0146dab2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5800,6 +5800,7 @@ enum { ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, + ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6850,6 +6851,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11040 }, + { 0x21, 0x04211020 }, + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7113,6 +7124,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -7280,6 +7292,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda - Apply AMD controller workaround for Raven platform
by Takashi Iwai 20 Sep '19

20 Sep '19

It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen Laptop) is fixed by the same workaround applied to other AMD chips. Update the driver_data entry for Raven (1022:15e3) to use the newly introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains AZX_DCAPS_PM_RUNTIME, we can drop that bit, too. Reported-and-tested-by: Dennis Padiernos <depadiernos(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..240f4ca76391 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = { AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.16.4

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[alsa-devel] alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Elimar Riesebieter 19 Sep '19

19 Sep '19

Hi ALSA devs, please notice the attached suggestion from Lenovo to support sound for whiskeylake sof-skl_hda_card audio. Thanks Elimar -- You cannot propel yourself forward by patting yourself on the back.

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[alsa-devel] [PATCH v1 0/9] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 19 Sep '19

19 Sep '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") Sylwester Nawrocki (9): ASoC: wm8994: Do not register inapplicable controls for WM1811 mfd: wm8994: Add support for MCLKn clock control ASoC: wm8994: Add support for setting MCLKn clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: arndale: Add missing OF node dereferencing ARM: dts: arndale: Add audio support (WM1811 CODEC boards) .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- drivers/mfd/wm8994-core.c | 9 + include/linux/mfd/wm8994/core.h | 9 + sound/soc/codecs/wm8994.c | 164 +++++++++++++++--- sound/soc/samsung/arndale_rt5631.c | 155 +++++++++++++---- 6 files changed, 306 insertions(+), 63 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v6] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 19 Sep '19

19 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

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[alsa-devel] DPCM: suspicious code in dpcm_prune_paths()
by Guennadi Liakhovetski 19 Sep '19

19 Sep '19

Hi, This code in dpcm_prune_paths() seems to have no effect: /* is there a valid CODEC DAI widget for this BE */ for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) continue; } Neither of the variables, that are modified in this loop are used outside of it. Was the "continue" maybe meant to continue the encompassing loop? Thanks Guennadi

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[alsa-devel] [PATCH v6 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Sep '19

19 Sep '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v5: - updated slave bindings with proper licencing and fixed up examples. Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + .../soundwire/soundwire-controller.yaml | 82 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1345 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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Re: [alsa-devel] [PATCH V2 3/4] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams
by S.j. Wang 19 Sep '19

19 Sep '19

Hi > > When set the runtime hardware parameters, we may need to query the > capability of DMA to complete the parameters. > > This patch is to Extract this operation from > dmaengine_pcm_set_runtime_hwparams function to a separate function > snd_dmaengine_pcm_set_runtime_hwparams, that other components > which need this feature can call this function. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > --- kbuild test robot report compile issue, will resend v3 after fixing. Best regards Wang shengjiu

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[alsa-devel] [PATCH] ASoC: core: delete component->card_list in soc_remove_component only
by Bard liao 19 Sep '19

19 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> We add component->card_list in the end of soc_probe_component(). In other words, component->card_list will not be added if there is an error in the soc_probe_component() function. So we can't delete component->card_list in the error handling of soc_probe_component(). Fixes: 22d1423187e5 ("ASoC: soc-core: add soc_cleanup_component()") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35f48e9c5ead..aff4b4bf4d07 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,7 +978,6 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); - list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -991,7 +990,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - + list_del(&component->card_list); soc_cleanup_component(component); } -- 2.17.1

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