June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/6] Add support of two Audio PLL source
by Shengjiu Wang 30 Jun '22

30 Jun '22

i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. The patches implement the functionality to select at runtime the appropriate AUDIO PLL for root clock, if there is no two PLL registered, then no action taken. Shengjiu Wang (6): ASoC: fsl_utils: Add function to handle PLL clock source ASoC: fsl_spdif: Add support for PLL switch at runtime. ASoC: fsl_micfil: Add support for PLL switch at runtime ASoC: fsl_sai: Add support for PLL switch at runtime ASoC: dt-bindings: fsl_spdif: Add two PLL clock source ASoC: dt-bindings: fsl-sai: Add two PLL clock source .../devicetree/bindings/sound/fsl,spdif.yaml | 4 ++ .../devicetree/bindings/sound/fsl-sai.txt | 3 + sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/fsl_micfil.c | 41 +++++++++++ sound/soc/fsl/fsl_sai.c | 54 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 2 + sound/soc/fsl/fsl_spdif.c | 57 +++++++++++++-- sound/soc/fsl/fsl_utils.c | 69 +++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 9 +++ 9 files changed, 237 insertions(+), 5 deletions(-) -- 2.17.1

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[PATCH v2] pinctrl: qcom: sc7280: Fix compile bug
by Srinivasa Rao Mandadapu 30 Jun '22

30 Jun '22

Fix the compilation error, caused by updating constant variable. Hence remove redundant constant variable, which is no more useful as per new design. The issue is due to some unstaged changes. Fix it up. Fixes: 36fe26843d6d ("pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Reviewed-by: Stephen Boyd <swboyd(a)chromium.org> --- Changes Since V1: -- Fix typo errors in commit message. drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 5 +---- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c index b5d1b99..e97ce45 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c @@ -388,9 +388,6 @@ int lpi_pinctrl_probe(struct platform_device *pdev) pctrl->data = data; pctrl->dev = &pdev->dev; - data->is_clk_optional = of_property_read_bool(dev->of_node, - "qcom,adsp-bypass-mode"); - pctrl->clks[0].id = "core"; pctrl->clks[1].id = "audio"; @@ -404,7 +401,7 @@ int lpi_pinctrl_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(pctrl->slew_base), "Slew resource not provided\n"); - if (data->is_clk_optional) + if (of_property_read_bool(dev->of_node, "qcom,adsp-bypass-mode")) ret = devm_clk_bulk_get_optional(dev, MAX_LPI_NUM_CLKS, pctrl->clks); else ret = devm_clk_bulk_get(dev, MAX_LPI_NUM_CLKS, pctrl->clks); diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h index 759d5d8..afbac2a 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h @@ -77,7 +77,6 @@ struct lpi_pinctrl_variant_data { int ngroups; const struct lpi_function *functions; int nfunctions; - bool is_clk_optional; }; int lpi_pinctrl_probe(struct platform_device *pdev); -- 2.7.4

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[PATCH] ASoC: samsung: change gpiod_speaker_power and rx1950_audio from global to static variables
by Tom Rix 30 Jun '22

30 Jun '22

sparse reports sound/soc/samsung/rx1950_uda1380.c:131:18: warning: symbol 'gpiod_speaker_power' was not declared. Should it be static? sound/soc/samsung/rx1950_uda1380.c:231:24: warning: symbol 'rx1950_audio' was not declared. Should it be static? Both gpiod_speaker_power and rx1950_audio are only used in rx1950_uda1380.c, so their storage class specifiers should be static. Fixes: 83d74e354200 ("ASoC: samsung: rx1950: turn into platform driver") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/samsung/rx1950_uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index ff3acc94a454..abf28321f7d7 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -128,7 +128,7 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } -struct gpio_desc *gpiod_speaker_power; +static struct gpio_desc *gpiod_speaker_power; static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -228,7 +228,7 @@ static int rx1950_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, &rx1950_asoc); } -struct platform_driver rx1950_audio = { +static struct platform_driver rx1950_audio = { .driver = { .name = "rx1950-audio", .pm = &snd_soc_pm_ops, -- 2.27.0

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[PATCH] ASoC: doc: Update dead links
by Marek Vasut 30 Jun '22

30 Jun '22

The alsa-project documentation is now part of the kernel docs, the original links are long dead, update links. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> --- Documentation/process/kernel-docs.rst | 2 +- Documentation/sound/soc/codec.rst | 2 +- Documentation/sound/soc/platform.rst | 2 +- sound/pci/ens1370.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/process/kernel-docs.rst b/Documentation/process/kernel-docs.rst index da9527502ef0e..644f9200fd919 100644 --- a/Documentation/process/kernel-docs.rst +++ b/Documentation/process/kernel-docs.rst @@ -108,7 +108,7 @@ On-line docs * Title: **Writing an ALSA Driver** :Author: Takashi Iwai <tiwai(a)suse.de> - :URL: http://www.alsa-project.org/~iwai/writing-an-alsa-driver/index.html + :URL: https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-dri… :Date: 2005 :Keywords: ALSA, sound, soundcard, driver, lowlevel, hardware. :Description: Advanced Linux Sound Architecture for developers, diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index 57df149acafc5..af973c4cac930 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -132,7 +132,7 @@ The codec driver also supports the following ALSA PCM operations:- }; Please refer to the ALSA driver PCM documentation for details. -http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ +https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html DAPM description diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst index c1badea53d3d3..7036630eaf016 100644 --- a/Documentation/sound/soc/platform.rst +++ b/Documentation/sound/soc/platform.rst @@ -46,7 +46,7 @@ snd_soc_component_driver:- }; Please refer to the ALSA driver documentation for details of audio DMA. -http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ +https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html An example DMA driver is soc/pxa/pxa2xx-pcm.c diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 94efe347a97a9..89210b2c73424 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -8,7 +8,7 @@ /* Power-Management-Code ( CONFIG_PM ) * for ens1371 only ( FIXME ) * derived from cs4281.c, atiixp.c and via82xx.c - * using http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ + * using https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-dri… * by Kurt J. Bosch */ -- 2.35.1

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Re: [PATCH] sound: delete a semicolon
by Mark Brown 30 Jun '22

30 Jun '22

On Thu, Jun 30, 2022 at 10:03:47AM +0800, Li kunyu wrote: > extra semicolons could be deleted. Please submit patches using subject lines reflecting the style for the subsystem, this makes it easier for people to identify relevant patches. Look at what existing commits in the area you're changing are doing and make sure your subject lines visually resemble what they're doing. There's no need to resubmit to fix this alone.

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[PATCH 0/2] ASoC: Intel: Skylake: Fixes for skl_get_ssp_clks()
by Peter Ujfalusi 30 Jun '22

30 Jun '22

Hi, while looking at long standing sparse reports regarding to arrays of flexible structures (arrays of flexible arrays of flexible structures, really). I came across these which did not looked right. The skl_get_ssp_clks() as some issues and can work under only strict single condition. Regards, Peter --- Peter Ujfalusi (2): ASoC: Intel: Skylake: Correct the ssp rate discovery in skl_get_ssp_clks() ASoC: Intel: Skylake: Correct the handling of fmt_config flexible array sound/soc/intel/skylake/skl-nhlt.c | 40 ++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 13 deletions(-) -- 2.37.0

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ucm2: add profile for the Librem 5
by GitHub pull_request - opened 30 Jun '22

30 Jun '22

alsa-project/alsa-ucm-conf pull request #181 was opened from craftyguy: cc @agx Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/181 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/181.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4 0/2] Read _SUB from ACPI to be able to identify firmware
by Stefan Binding 29 Jun '22

29 Jun '22

CS35L41 has a DSP which is able to run firmware, as well as a tuning file. Different systems may want to use different firmwares and tuning files, and some firmwares/tunings may not be compatible with other systems. To allow a system to select the correct fimware/tuning, we can read an _SUB from the ACPI. This _SUB can then be used to uniquely identify the system in the firmware/tuning file name. Add a helper function which reads the _SUB, so this can be used by other parts in the future. Add support inside the CS35L41 ASoC driver to read this _SUB, and save it appropriately. Changes since v3: - Fix 32 bit format string warning Changes since v2: - Fix error in function prototype Changes since v1: - Add length validation for SSID String - Rename API - Allocate memory inside API - Use ACPI_HANDLE macro instead of ACPI_COMPANION - Improve error handling Stefan Binding (2): ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware drivers/acpi/utils.c | 38 ++++++++++++++++++++++++++++++++++++++ include/linux/acpi.h | 6 ++++++ sound/soc/codecs/cs35l41.c | 24 ++++++++++++++++++++++++ 3 files changed, 68 insertions(+) -- 2.25.1

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[PATCH 0/4] ASoC: codecs: add WSA883x support
by Srinivas Kandagatla 29 Jun '22

29 Jun '22

This patchset adds support for WSA883x smart speaker amplifier codec connected via SoundWire. This codec also has a temperature sensor used for speaker protection, support for this is not added yet. Most of the code is derived from Qualcomm downstream msm-5.10 kernel. Thanks to Patrick Lai's Team. This codec is tested on SM8450 MTP. Thanks, Srini Srinivas Kandagatla (4): ASoC: dt-bindings: Add WSA883x bindings ASoC: codecs: add wsa883x amplifier support ASoC: codecs: wsa883x: add control, dapm widgets and map MAINTAINERS: add ASoC Qualcomm codecs .../bindings/sound/qcom,wsa883x.yaml | 74 + MAINTAINERS | 4 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa883x.c | 1498 +++++++++++++++++ 5 files changed, 1588 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml create mode 100644 sound/soc/codecs/wsa883x.c -- 2.25.1

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[PATCH] ASoC: wcd9335: Use int array instead of bitmask for TX mixers
by Yassine Oudjana 29 Jun '22

29 Jun '22

From: Yassine Oudjana <y.oudjana(a)protonmail.com> Currently slim_tx_mixer_get reports all TX mixers as enabled when at least one is, due to it reading the entire tx_port_value bitmask without testing the specific bit corresponding to a TX port. Furthermore, using the same bitmask for all capture DAIs makes setting one mixer affect them all. To prevent this, and since the SLIM TX muxes effectively only connect to one of the mixers at a time, turn tx_port_value into an int array storing the DAI index each of the ports is connected to. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/codecs/wcd9335.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 7d40a61b03b0..a4d92f7aef9e 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -342,7 +342,7 @@ struct wcd9335_codec { struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; - unsigned int tx_port_value; + unsigned int tx_port_value[WCD9335_TX_MAX]; int hph_l_gain; int hph_r_gain; u32 rx_bias_count; @@ -1334,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int dai_id = widget->shift; + int port_id = mixer->shift; - ucontrol->value.integer.value[0] = wcd->tx_port_value; + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id; return 0; } @@ -1358,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, case AIF2_CAP: case AIF3_CAP: /* only add to the list if value not set */ - if (enable && !(wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value |= BIT(port_id); + if (enable && wcd->tx_port_value[port_id] != dai_id) { + wcd->tx_port_value[port_id] = dai_id; list_add_tail(&wcd->tx_chs[port_id].list, &wcd->dai[dai_id].slim_ch_list); - } else if (!enable && (wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value &= ~BIT(port_id); + } else if (!enable && wcd->tx_port_value[port_id] == dai_id) { + wcd->tx_port_value[port_id] = -1; list_del_init(&wcd->tx_chs[port_id].list); } break; -- 2.36.1

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