February 2023 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH 1/2] ALSA: cs35l41: Add shared boost feature
by Charles Keepax 07 Feb '23

07 Feb '23

On Tue, Feb 07, 2023 at 10:40:20AM +0000, Lucas Tanure wrote: > Shared boost allows two amplifiers to share a single boost > circuit by communicating on the MDSYNC bus. > The passive amplifier does not control the boost and receives > data from the active amplifier. > > Shared Boost is not supported in HDA Systems. > Probably would be nice to put at least a note to say based on David's patches. > +static const struct reg_sequence cs35l41_shd_boost_seq[] = { > + {CS35L41_PWR_CTRL3, 0x01000110}, This will blat whatever the user set in the DRE switch. Technically blats the CLASS H enable from the DAPM widget too, but as that always turns on should be a no-op. Probably should either not register the DRE switch or have setting it return an error for these boost modes. > +int cs35l41_global_enable(struct regmap *regmap, enum cs35l41_boost_type b_type, int enable, > + struct completion *pll_lock) > { > int ret; > + unsigned int gpio1; > > switch (b_type) { > + case CS35L41_SHD_BOOST_ACTV: > + case CS35L41_SHD_BOOST_PASS: > + regmap_update_bits(regmap, CS35L41_PWR_CTRL3, CS35L41_SYNC_EN_MASK, 0); > + > + gpio1 = enable ? CS35L41_GPIO1_MDSYNC : CS35L41_GPIO1_HIZ; > + regmap_update_bits(regmap, CS35L41_GPIO_PAD_CONTROL, CS35L41_GPIO1_CTRL_MASK, > + gpio1 << CS35L41_GPIO1_CTRL_SHIFT); > + > + ret = regmap_update_bits(regmap, CS35L41_PWR_CTRL1, CS35L41_GLOBAL_EN_MASK, > + enable << CS35L41_GLOBAL_EN_SHIFT); > + usleep_range(3000, 3100); > + if (!enable) > + break; > + > + if (!pll_lock) > + return -EINVAL; > + > + ret = wait_for_completion_timeout(pll_lock, msecs_to_jiffies(1000)); > + if (ret == 0) > + ret = -ETIMEDOUT; This feels kinda scary, in that you are relying on a 1 to 1 correspondence between this code running and getting a PLL lock signal. The datasheet is helpfully completely vague on when PLL locks are triggered. The PLL enable seems to be set through set_sysclk, which could be called multiple times, per DAPM power up. Does the PLL lock only go once global enable has been set? Can't help but wonder if a reinit_completion should probably go somewhere to ensure we are getting this lock of the PLL not a past one. > @@ -483,6 +483,11 @@ static irqreturn_t cs35l41_irq(int irq, void *data) > ret = IRQ_HANDLED; > } > > + if (status[2] & CS35L41_PLL_LOCK) { > + regmap_write(cs35l41->regmap, CS35L41_IRQ1_STATUS3, CS35L41_PLL_LOCK); > + complete(&cs35l41->pll_lock); > + } > + If you fall into any of the error cases in this IRQ handler above this, it will blat values you don't want into BST_EN although, to be fair that does look currently broken for external boost as well. Thanks, Charles

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[PATCH] ASoC: topology: Return -ENOMEM on memory allocation failure
by Amadeusz Sławiński 07 Feb '23

07 Feb '23

When handling error path, ret needs to be set to correct value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Fixes: d29d41e28eea ("ASoC: topology: Add support for multiple kcontrol types to a widget") Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 78223603088e..07421f5d4ebd 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1369,13 +1369,17 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.num_kcontrols = le32_to_cpu(w->num_kcontrols); kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); - if (!kc) + if (!kc) { + ret = -ENOMEM; goto hdr_err; + } kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), GFP_KERNEL); - if (!kcontrol_type) + if (!kcontrol_type) { + ret = -ENOMEM; goto hdr_err; + } for (i = 0; i < le32_to_cpu(w->num_kcontrols); i++) { control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; -- 2.34.1

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[PATCH v2 0/7] ASoC: dt-bindings: meson: covert axg audio to schema
by Jerome Brunet 07 Feb '23

07 Feb '23

Convert AXG audio dt-binding documentation to schema Changes since v1: * Drop 2 patches applied with v1 * Drop node name patterns * Fix examples indentation * Yaml fixups requested by Krzysztof Jerome Brunet (7): ASoC: dt-bindings: meson: convert axg tdm interface to schema ASoC: dt-bindings: meson: convert axg tdm formatters to schema ASoC: dt-bindings: meson: convert axg pdm to schema ASoC: dt-bindings: meson: convert axg fifo to schema ASoC: dt-bindings: meson: convert axg spdif input to schema ASoC: dt-bindings: meson: convert axg spdif output to schema ASoC: dt-bindings: meson: convert axg sound card control to schema .../bindings/sound/amlogic,axg-fifo.txt | 34 ---- .../bindings/sound/amlogic,axg-fifo.yaml | 112 +++++++++++ .../bindings/sound/amlogic,axg-pdm.txt | 29 --- .../bindings/sound/amlogic,axg-pdm.yaml | 82 ++++++++ .../bindings/sound/amlogic,axg-sound-card.txt | 124 ------------ .../sound/amlogic,axg-sound-card.yaml | 183 ++++++++++++++++++ .../bindings/sound/amlogic,axg-spdifin.txt | 27 --- .../bindings/sound/amlogic,axg-spdifin.yaml | 86 ++++++++ .../bindings/sound/amlogic,axg-spdifout.txt | 25 --- .../bindings/sound/amlogic,axg-spdifout.yaml | 79 ++++++++ .../sound/amlogic,axg-tdm-formatters.txt | 36 ---- .../sound/amlogic,axg-tdm-formatters.yaml | 88 +++++++++ .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 --- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 55 ++++++ 14 files changed, 685 insertions(+), 297 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml -- 2.39.0

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[PATCH] ASoC: fsl_sai: fix getting version from VERID
by Shengjiu Wang 07 Feb '23

07 Feb '23

The version information is at the bit31 ~ bit16 in the VERID register, so need to right shift 16bit to get it, otherwise the result of comparison "sai->verid.version >= 0x0301" is wrong. Fixes: 99c1e74f25d4 ("ASoC: fsl_sai: store full version instead of major/minor") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c365afd6c4ea..1b197478b3d9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1142,6 +1142,7 @@ static int fsl_sai_check_version(struct device *dev) sai->verid.version = val & (FSL_SAI_VERID_MAJOR_MASK | FSL_SAI_VERID_MINOR_MASK); + sai->verid.version >>= FSL_SAI_VERID_MINOR_SHIFT; sai->verid.feature = val & FSL_SAI_VERID_FEATURE_MASK; ret = regmap_read(sai->regmap, FSL_SAI_PARAM, &val); -- 2.34.1

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Re: Firewire Phase24 Motu help
by Takashi Sakamoto 07 Feb '23

07 Feb '23

Hi, (C.C.ed to administrator of alsa-devel mailing list) On Mon, Jan 30, 2023 at 12:40:06PM +0100, M. Armsby wrote: > Hi Takashi, > I'm writing direct because I cannot seem to get a post on AlsaDevel > list. I've realized your two posts, but no reaction since they have no message body: * Firewire ALSA Motu and Phase24 * https://lore.kernel.org/alsa-devel/sokos1-r6xj933yn42y-hmy3gsrfgsma-mnwe9a-… * Firewire Motu ALSA (Re-post) * https://lore.kernel.org/alsa-devel/v6inx6-fwmr0r-kvajcd-kfs8u9l6m2qn2cnogd-… Jaroslav, would I ask your opinion for the problem which perhaps dropped the message body from his messages? > I need to repair my two Phase X24 (defekt caps?) and I'm having > problems with Traveler and other Motu interfaces on Linux. > Thanks for any hints or where I should post. > Here's a copy of my post: > Hello everyone! > Thank you for all the hard work on this. > Especially Mr. Takashi Sakamoto. > I'm an experienced studio engineer and was a Beta tester for Samplitude > for years. > Not a programmer and have very little experience with Linux. > So please excuse me blundering in on this professional programmer forum > but I'm hoping my experience will help somehow to find bugs. I'll > provide any information I can or test following instructions. > I really would love to completely change over to Linux but I need > reliable drivers. > How can I help? > Here's my experience so far. > Around November I started testing Linux for audio as I have > occasionally over the last years. I was pleased to find Manjaro and > Sparky would detect Traveler Mk1 and Mk3 and to run so efficiently > compared to Win10. > But things would freeze. > Reaper has a long standing Midi bug so it was difficult at first to see > why my laptop would freeze on occasion. I dissabled hardware midi in > the software. > Then I updated Sparky beginning of December and to my surprise Firewire > didn't work anymore. The Firewire stack must have been removed. > I use: > Dell Laptops 4310, 6510, M4800 > Tested platforms Ardour and Reaper using Pianoteq 8 > No JACK > Manjaro Kernel from 6.0 rc4? and 6.1.1 work for a limited time. > Sparky only works with special kernels Xanmod1 6.1.7-x64v2 but the > audio is completely distorted. > CachyOS Kernel 6.2 rc4 sounds good. > Common problem is freezing or loosing audio. Ardour is less sensitive > than Reaper but Reaper is far more efficient on CPU > (20% compared to 90%and more with one instance of Pianoteq 7 or 8) > Motu Traveler MK3 and MK1 wrong channel assignment from 1+2 (3+4 are > actually 1+2) and freezing or just loosing audio. > Motu 896Mk3 Hybrid not recognised in firewire or USB mode. > Shame cause its the best. > Motu 8pre working similarly. > Phase 24 working much better but still freezes or looses audio after > long period. > Tested in Ardour and Reaper. > I hope that's not too much at once. > Martin Armsby > M. Armsby Thanks for the report including many issues. I apologize to trouble you. In my opinion, the issues are not so easy since each application uses ALSA PCM device by slightly different ways. So we need to investigate the issue step by step to take relevant longer time than easy issues. Before discussing about the issue, I need to ask a help about MOTU devices. As you realized, 896 mk3 is neither supported yet, FireWire only nor Hybrid, in IEEE 1394 bus nor Universal Serial Bus. For example, you can see no entry for the device in module device table[1] in the file which consists of snd-firewire-motu driver. The reason is apparent; no one provides information enough to support it. Then I would like you to help me to integrate for it in the case in IEEE 1394 bus. I maintain a remote repository for my collection of configuration ROM image which includes identity information of each node in IEEE 1394 bus[2]. The information is useful both to make the entry of module device table and to make hardware database[3] in systemd project. When 896 mk3 Hybrid is connected to IEEE 1394 bus and only connected to the bus, Linux FireWire subsystem detects it as '/dev/fw1'. Then the rom image is exported to an node on sysfs. You can retrieve the content by the command line below: $ cat /sys/bus/firewire/devices/fw1/config_rom > motu-896mk3hybrid.img If retrieved successfully, please send it to me. I don't mind to receive it to private message, mailing list, any upload services. [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… [2] https://github.com/takaswie/am-config-roms [3] https://github.com/systemd/systemd/blob/main/hwdb.d/80-ieee1394-unit-functi… Thanks Takashi Sakamoto

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[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform.
by Andy Chi 07 Feb '23

07 Feb '23

There is a HP platform needs ALC236_FIXUP_HP_GPIO_LED quirk to make mic-mute/audio-mute working. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4055a8f5880a..d1258dd803e8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9433,6 +9433,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.34.1

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[PATCH] soundwire: export sdw_compute_slave_ports() function
by Vijendar Mukunda 07 Feb '23

07 Feb '23

Export sdw_compute_slave_ports() function to use it in another soundwire manager module. Move sdw_transport_data structure to bus header file to export sdw_compute_slave_ports() function. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/lkml/20230201165944.3169125-1-Vijendar.Mukunda@amd.… --- drivers/soundwire/bus.h | 9 +++++++++ drivers/soundwire/generic_bandwidth_allocation.c | 12 +++--------- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 7631ef5e71fb..141b99ac58de 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -144,6 +144,13 @@ struct sdw_master_runtime { struct list_head bus_node; }; +struct sdw_transport_data { + int hstart; + int hstop; + int block_offset; + int sub_block_offset; +}; + struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, enum sdw_data_direction direction, unsigned int port_num); @@ -213,5 +220,7 @@ int sdw_bwrite_no_pm_unlocked(struct sdw_bus *bus, u16 dev_num, u32 addr, u8 val void sdw_clear_slave_status(struct sdw_bus *bus, u32 request); int sdw_slave_modalias(const struct sdw_slave *slave, char *buf, size_t size); +void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, + struct sdw_transport_data *t_data); #endif /* __SDW_BUS_H */ diff --git a/drivers/soundwire/generic_bandwidth_allocation.c b/drivers/soundwire/generic_bandwidth_allocation.c index f7c66083a4dd..39543048baa7 100644 --- a/drivers/soundwire/generic_bandwidth_allocation.c +++ b/drivers/soundwire/generic_bandwidth_allocation.c @@ -28,15 +28,8 @@ struct sdw_group { unsigned int *rates; }; -struct sdw_transport_data { - int hstart; - int hstop; - int block_offset; - int sub_block_offset; -}; - -static void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, - struct sdw_transport_data *t_data) +void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, + struct sdw_transport_data *t_data) { struct sdw_slave_runtime *s_rt = NULL; struct sdw_port_runtime *p_rt; @@ -85,6 +78,7 @@ static void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, } } } +EXPORT_SYMBOL(sdw_compute_slave_ports); static void sdw_compute_master_ports(struct sdw_master_runtime *m_rt, struct sdw_group_params *params, -- 2.34.1

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[PATCH] ASoC: dt-bindings: renesas: adjust to R-Car Gen4
by Kuninori Morimoto 07 Feb '23

07 Feb '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car Gen4 is not compatible with Gen3, this patch adjusts to R-Car Gen4. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- The "required" with if - then - else on "rcar_sound,ssi" is always match to "then" even though it is checking "renesas,rcar_sound-gen4" or not. Why ?? Is it my fault ?? .../bindings/sound/renesas,rsnd.yaml | 62 ++++++++++++++----- 1 file changed, 46 insertions(+), 16 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index d106de00c6b2..9a88b1c34e72 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -106,7 +106,9 @@ properties: items: oneOf: - const: ssi-all + - const: clkin - pattern: '^ssi\.[0-9]$' + - pattern: '^ssiu\.[0-9]$' - pattern: '^src\.[0-9]$' - pattern: '^mix\.[0-1]$' - pattern: '^ctu\.[0-1]$' @@ -254,10 +256,20 @@ properties: no-busif: description: BUSIF is not used when [mem -> SSI] via DMA case $ref: /schemas/types.yaml#/definitions/flag - required: - - interrupts - - dmas - - dma-names + allOf: + - if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + required: + - interrupts + else: + required: + - interrupts + - dmas + - dma-names additionalProperties: false # For DAI base @@ -307,18 +319,36 @@ allOf: - ssi - adg else: - properties: - reg: - maxItems: 5 - reg-names: - maxItems: 5 - items: - enum: - - scu - - adg - - ssiu - - ssi - - audmapp + if: + properties: + compatible: + contains: + const: renesas,rcar_sound-gen4 + then: + properties: + reg: + maxItems: 4 + reg-names: + maxItems: 4 + items: + enum: + - adg + - ssiu + - ssi + - sdmc + else: + properties: + reg: + maxItems: 5 + reg-names: + maxItems: 5 + items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp unevaluatedProperties: false -- 2.25.1

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[PATCH] ASoC: codecs: es8326: Fix DTS properties reading
by Alexey Firago 06 Feb '23

06 Feb '23

Seems like properties parsing and reading was copy-pasted, so "everest,interrupt-src" and "everest,interrupt-clk" are saved into the es8326->jack_pol variable. This might lead to wrong settings being saved into the reg 57 (ES8326_HP_DET). Fix this by using proper variables while reading properties. Signed-off-by: Alexey Firago <a.firago(a)yadro.com> --- sound/soc/codecs/es8326.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 9ddf6a35e91c..28a0565c2a95 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -729,14 +729,16 @@ static int es8326_probe(struct snd_soc_component *component) } dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); - ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-src", + &es8326->interrupt_src); if (ret != 0) { dev_dbg(component->dev, "interrupt-src return %d", ret); es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; } dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); - ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", + &es8326->interrupt_clk); if (ret != 0) { dev_dbg(component->dev, "interrupt-clk return %d", ret); es8326->interrupt_clk = 0x45; -- 2.34.1

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[PATCH v3 0/2] Two bug fixes for tas5805m codec driver
by Daniel Beer 06 Feb '23

06 Feb '23

This pair of patches fixes two issues which crept in while revising the original submission, at a time when I no longer had access to test hardware. The fixes here have been tested and verified on hardware. Changes v2 -> v3: - Resubmit with cover letter Changes v1 -> v2: - Fix issue reported by kernel test robot Daniel Beer (2): ASoC: tas5805m: rework to avoid scheduling while atomic. ASoC: tas5805m: add missing page switch. sound/soc/codecs/tas5805m.c | 131 +++++++++++++++++++++++++----------- 1 file changed, 90 insertions(+), 41 deletions(-) -- 2.38.1

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