July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: soc-core: Fix regression causing sysfs entries to disappear
by Charles Keepax 31 Jul '20

31 Jul '20

The allocation order of things in soc_new_pcm_runtime was changed to move the device_register before the allocation of the rtd structure. This was to allow the rtd allocation to be managed by devm. However currently the sysfs entries are added by device_register and their visibility depends on variables within the rtd structure, this causes the pmdown_time and dapm_widgets sysfs entries to be missing for all rtds. Correct this issue by manually calling device_add_groups after the appropriate information is available. Fixes: d918a37610b1 ("ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index defd96b14c287..df4c7116f308c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -446,7 +446,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( dev->parent = card->dev; dev->release = soc_release_rtd_dev; - dev->groups = soc_dev_attr_groups; dev_set_name(dev, "%s", dai_link->name); @@ -503,6 +502,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( /* see for_each_card_rtds */ list_add_tail(&rtd->list, &card->rtd_list); + ret = device_add_groups(dev, soc_dev_attr_groups); + if (ret < 0) + goto free_rtd; + return rtd; free_rtd: -- 2.11.0

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[PATCH v4] ASoC: dt-bindings: ak4613: switch to yaml base Documentation
by Kuninori Morimoto 31 Jul '20

31 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/87mu4cxlo2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o8pf3923.wl-kuninori.morimoto.gx@renesas.com --- v3 -> v4 - rebase v2 patch - use generic node name Hi Mark, Rob I posted ak4613 3 times (= v1 - v3), and v2 patch got Reviewed-by from Rob (= https://lore.kernel.org/r/20200715220920.GA903738@bogus). But in the same time, it got review from Fabio to use generic node name. Then, I used v1 for v3 somehow and everything becoming wrong. This patch should be OK. I added Rob's Reviewed-by on this patch, too. .../devicetree/bindings/sound/ak4613.txt | 27 ---------- .../devicetree/bindings/sound/ak4613.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 27 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4613.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt deleted file mode 100644 index 49a2e74fd9cb..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ /dev/null @@ -1,27 +0,0 @@ -AK4613 I2C transmitter - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak4613" -- reg : The chip select number on the I2C bus - -Optional properties: -- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- asahi-kasei,in2-single-end rather than differential. -- asahi-kasei,out1-single-end -- asahi-kasei,out2-single-end -- asahi-kasei,out3-single-end -- asahi-kasei,out4-single-end -- asahi-kasei,out5-single-end -- asahi-kasei,out6-single-end - -Example: - -&i2c { - ak4613: ak4613@10 { - compatible = "asahi-kasei,ak4613"; - reg = <0x10>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000000..ef4055ef0ccd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; -- 2.25.1

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[PATCH 0/4] ASoC: meson: tdm fixes
by Jerome Brunet 31 Jul '20

31 Jul '20

This patcheset is collection of fixes for the TDM input and output the axg audio architecture. Its fixes: - slave mode format setting - g12 and sm1 skew offset - tdm clock inversion - standard daifmt props names which don't require a specific prefix Jerome Brunet (4): ASoC: meson: axg-tdm-interface: fix link fmt setup ASoC: meson: axg-tdmin: fix g12a skew ASoC: meson: axg-tdm-formatters: fix sclk inversion ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties sound/soc/meson/axg-tdm-formatter.c | 11 ++++++----- sound/soc/meson/axg-tdm-formatter.h | 1 - sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++--------- sound/soc/meson/axg-tdmin.c | 16 +++++++++++++++- sound/soc/meson/axg-tdmout.c | 3 --- sound/soc/meson/meson-card-utils.c | 2 +- 6 files changed, 39 insertions(+), 20 deletions(-) -- 2.25.4

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[PATCH] ASoC: SOF: imx: use resource_size
by Julia Lawall 31 Jul '20

31 Jul '20

Use resource_size rather than a verbose computation on the end and start fields. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) <smpl> @@ struct resource ptr; @@ - (ptr.end - ptr.start + 1) + resource_size(&ptr) </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/sof/imx/imx8m.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff -u -p a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -188,8 +188,7 @@ static int imx8m_probe(struct snd_sof_de } sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start, - res.end - res.start + - 1); + resource_size(&res)); if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n", base, size);

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[PATCH 1/2] ASoC: tlv320adcx140: Fix GPO register start address
by Dan Murphy 31 Jul '20

31 Jul '20

The header was updated to align with the data sheet to start the GPO_CFG at GPO_CFG0. The code was not updated to the change and therefore the GPO_CFG0 register was not written to. Fixes: 6617cff6a05e ("ASoC: tlv320adcx140: Add GPO configuration and drive output config") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 7bb9e9aa0c0a..fc7616dcf9b5 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -790,7 +790,7 @@ static int adcx140_configure_gpo(struct adcx140_priv *adcx140) gpo_output_val = gpo_outputs[0] << ADCX140_GPO_SHIFT | gpo_outputs[1]; - ret = regmap_write(adcx140->regmap, ADCX140_GPO_CFG1 + i, + ret = regmap_write(adcx140->regmap, ADCX140_GPO_CFG0 + i, gpo_output_val); if (ret) return ret; -- 2.28.0

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[PATCH v2 1/5] ASoC: amd: Renaming snd-soc-card structure and fields
by Ravulapati Vishnu vardhan rao 31 Jul '20

31 Jul '20

As in future our machine driver supports multiple codecs So changing naming convention of snd_soc_card struct and its fields. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 6009e444b858..8b5af064864f 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -290,7 +290,7 @@ static const struct snd_kcontrol_new acp3x_dmic_mux_control = SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, dmic_get, dmic_set); -static const struct snd_soc_dapm_widget acp3x_widgets[] = { +static const struct snd_soc_dapm_widget acp3x_5682_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -298,7 +298,7 @@ static const struct snd_soc_dapm_widget acp3x_widgets[] = { &acp3x_dmic_mux_control), }; -static const struct snd_soc_dapm_route acp3x_audio_route[] = { +static const struct snd_soc_dapm_route acp3x_5682_audio_route[] = { {"Headphone Jack", NULL, "HPOL"}, {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, @@ -307,23 +307,23 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Dmic Mux", "Rear Mic", "DMIC"}, }; -static const struct snd_kcontrol_new acp3x_mc_controls[] = { +static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Spk"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; -static struct snd_soc_card acp3x_card = { +static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, .dai_link = acp3x_dai_5682_98357, .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), - .dapm_widgets = acp3x_widgets, - .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), - .dapm_routes = acp3x_audio_route, - .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), - .controls = acp3x_mc_controls, - .num_controls = ARRAY_SIZE(acp3x_mc_controls), + .dapm_widgets = acp3x_5682_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), + .dapm_routes = acp3x_5682_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_5682_audio_route), + .controls = acp3x_5682_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; static int acp3x_probe(struct platform_device *pdev) @@ -336,8 +336,8 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; - card = &acp3x_card; - acp3x_card.dev = &pdev->dev; + card = &acp3x_5682; + acp3x_5682.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,11 +348,11 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_card.name, ret); + acp3x_5682.name, ret); return ret; } return 0; -- 2.17.1

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[PATCH] ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection
by Alper Nebi Yasak 31 Jul '20

31 Jul '20

PulseAudio (and perhaps other userspace utilities) can not detect any jack for rk3399_gru_sound as the driver doesn't expose related Jack kcontrols. This patch adds two DAPM pins to the headset jack, where the snd_soc_card_jack_new() call automatically creates "Headphones Jack" and "Headset Mic Jack" kcontrols from them. With an appropriate ALSA UCM config specifying JackControl fields for the "Headphones" and "Headset" (mic) devices, PulseAudio can detect plug/unplug events for both of them after this patch. Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b30..c17d146717a8 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -32,6 +32,19 @@ static unsigned int dmic_wakeup_delay; static struct snd_soc_jack rockchip_sound_jack; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin rockchip_sound_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + +}; + static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -176,7 +189,9 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &rockchip_sound_jack, NULL, 0); + &rockchip_sound_jack, + rockchip_sound_jack_pins, + ARRAY_SIZE(rockchip_sound_jack_pins)); if (ret) { dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); -- 2.28.0.rc1

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[PATCH] ASoC: da7219: Fix general protection fault in da7219_register_dai_clks
by Yong Zhi 30 Jul '20

30 Jul '20

clkdev_drop(cl) does not null the removed cl, if da7219_register_dai_clks() entered again after card removal, devm_clk_register() will return -EEXIST, the goto err to clkdev_drop() will trigger board reboot. Test commands: modprobe -r snd_sof_pci modprobe snd_sof_pci The oops looks like: da7219 i2c-DLGS7219:00: Using default DAI clk names: da7219-dai-wclk, da7219-dai-bclk da7219 i2c-DLGS7219:00: Failed to register da7219-dai-wclk: -17 general protection fault: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:clkdev_drop+0x20/0x52 Call Trace: da7219_probe+0x52e/0x6dc [snd_soc_da7219] soc_probe_component+0x206/0x3a1 snd_soc_bind_card+0x4ee/0x9a6 devm_snd_soc_register_card+0x48/0x7b audio_probe+0x1f0/0x221 [snd_soc_sof_da7219_max98373] platform_drv_probe+0x89/0xa2 really_probe+0x129/0x30d driver_probe_device+0x59/0xec ? driver_sysfs_remove+0x55/0x55 bus_for_each_drv+0xa1/0xdc __device_attach+0xc2/0x146 bus_probe_device+0x32/0x97 device_add+0x311/0x3b4 platform_device_add+0x184/0x1eb Fix by marking (nullifying) the da7219->dai_clks_lookup[i] after clkdev_drop(). Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/da7219.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 153ea30b5a8f..54da7cfbb5f4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2369,8 +2369,10 @@ static void da7219_remove(struct snd_soc_component *component) #ifdef CONFIG_COMMON_CLK for (i = DA7219_DAI_NUM_CLKS - 1; i >= 0; --i) { - if (da7219->dai_clks_lookup[i]) + if (da7219->dai_clks_lookup[i]) { clkdev_drop(da7219->dai_clks_lookup[i]); + da7219->dai_clks_lookup[i] = NULL; + } } #endif -- 2.7.4

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[PATCH] ALSA: hda: fix NULL pointer dereference during suspend
by Ranjani Sridharan 30 Jul '20

30 Jul '20

When the ASoC card registration fails and the codec component driver never probes, the codec device is not initialized and therefore memory for codec->wcaps is not allocated. This results in a NULL pointer dereference when the codec driver suspend callback is invoked during system suspend. Fix this by returning without performing any actions during codec suspend/resume if the card was not registered successfully. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3576e2d8452f..9b1f387d18e5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2936,6 +2936,10 @@ static int hda_codec_runtime_suspend(struct device *dev) struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; + /* Nothing to do if card registration fails and the component driver never probes */ + if (!codec->card) + return 0; + cancel_delayed_work_sync(&codec->jackpoll_work); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || @@ -2950,6 +2954,10 @@ static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* Nothing to do if card registration fails and the component driver never probes */ + if (!codec->card) + return 0; + codec_display_power(codec, true); snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); -- 2.25.1

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[PATCH v2 0/2] Add period size constraint for Atom Chromebook
by Brent Lu 30 Jul '20

30 Jul '20

Two different constraints are implemented: one is in platform's CPU DAI to enforce period sizes which are already used in Android BSP. The other is in Atom Chromebook's machine driver to use 240 as period size. Changes since v1: -Add comma at the end of media_period_size array declaration. Brent Lu (1): ASoC: intel: atom: Add period size constraint Yu-Hsuan Hsu (1): ASoC: Intel: Add period size constraint on strago board sound/soc/intel/atom/sst-mfld-platform-pcm.c | 15 +++++++++++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 +++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++++++++++++- 3 files changed, 41 insertions(+), 2 deletions(-) -- 2.7.4

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