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Invalid jack configuration on Mac Pro 2013
by Julio Merino 24 Sep '23

24 Sep '23

Hello, I have a Mac Pro 2013 running Fedora 38 and, while I get sound, the jack connections seem to be incorrectly configured. I tried to follow https://docs.kernel.org/sound/hd-audio/notes.html for troubleshooting, and went the route of trying to figure out pin configuration changes... but so far haven't succeeded. So... I'm sending this as a bug report per the instructions in that document. Here is what I observe: - The machine has two jacks: line out and headphones. - The line out jack emits red light while playing. - When nothing is connected to either jack, sound plays through the internal speaker: OK. - When connecting speakers to the line out jack, the internal speaker is muted but no sound comes through the speakers. - When connecting headphones to the headphones jack, sound comes through the headphones BUT this only happens when something /else/ is connected to the line out jack to silence the internal speaker. Otherwise the headphones jack seems non-functional. - The volume indicator in Gnome switches from speaker to headphones when something is connected to the line out jack. It doesn't recognize any changes when connecting something to the headphones jack. - I tried booting with model=mbp11 and the only thing that seems to change is that the red light turns off. Here is the output of alsa-info.sh: http://alsa-project.org/db/?f=07112f6bc87c0938b38394e7d52aeecb307afcbe Please let me know if you need any other details. Happy to test out any changes via a "patch firmware" file or sysfs settings. Thank you

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mainline build failure due to ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver")
by Sudip Mukherjee (Codethink) 24 Sep '23

24 Sep '23

Hi All, The latest mainline kernel branch fails to build alpha, csky and s390 allmodconfig with the error: drivers/mfd/cs42l43.c:1138:12: error: 'cs42l43_runtime_resume' defined but not used [-Werror=unused-function] 1138 | static int cs42l43_runtime_resume(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~ drivers/mfd/cs42l43.c:1124:12: error: 'cs42l43_runtime_suspend' defined but not used [-Werror=unused-function] 1124 | static int cs42l43_runtime_suspend(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~ drivers/mfd/cs42l43.c:1106:12: error: 'cs42l43_resume' defined but not used [-Werror=unused-function] 1106 | static int cs42l43_resume(struct device *dev) | ^~~~~~~~~~~~~~ drivers/mfd/cs42l43.c:1076:12: error: 'cs42l43_suspend' defined but not used [-Werror=unused-function] 1076 | static int cs42l43_suspend(struct device *dev) | ^~~~~~~~~~~~~~~ git bisect pointed to ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver") I will be happy to test any patch or provide any extra log if needed. -- Regards Sudip

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Fwd: Re: MAN301 external clock issues (Re: [PATCH] ALSA: dice: add stream format parameters for Weiss devices)
by Michele Perrone 24 Sep '23

24 Sep '23

Hi Takashi, On 19/09/23 16:09, Takashi Sakamoto wrote: > For the case of external source of clock, you need to configure the > sampling rate in device by your hand, by accesing to GLOBAL_CLOCK_SELECT > offset in global section of TCAT protocol. I'm able to do this from the kernel space, but I wouldn't know how to achieve this in the user-space without switching back and forth between internal and external clock source. Moreover, there seems to be no 'write' handler installed inside 'snd_dice_create_hwdep()'[2], so using 'snd_hwdep_write()' from user-space ALSA is not possible. The only option I see is adding a new control to 'snd_kcontrol_new controls[]'[3], inside 'dice_snd_ctl_construct()'[3] in our AV/C code. But I don't think this is the best solution, in view of the fact the we will be switching to snd-firewire-ctl-services[4] in the near future. > I programmed ALSA dice driver to behave like that. > > The driver configures the target device for sampling rate which ALSA PCM > application expects when the device uses internal source of clock. When > the device is configured to use external source of clock, the driver > adds constrains to ALSA PCM hardware configuration so that the application > works with current sampling rate configured in device. Thanks for explaining this. I have now found the place in the code where you do this[1]. May I ask, what is the reason for this limitation? I don't see any relevant connection between that last used internal sampling rate, and the current external sampling rate. Moreover, the DICE driver is able to see the external sampling rate in real-time. So when the external clock rate is selected and locked, wouldn't it make sense that the device rate followed the external rate automatically, without user-space intervention? > Regards > > Takashi Sakamoto Kind regards, Michele Perrone [1]https://github.com/torvalds/linux/blob/v6.6-rc2/sound/firewire/dice/dice-… [2]https://github.com/torvalds/linux/blob/v6.6-rc2/sound/firewire/dice/dice-… [3]https://github.com/weiss-engineering/snd-dice/blob/5a95496c0666048bc5bc8c… [4]https://github.com/alsa-project/snd-firewire-ctl-services

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[PATCH] ALSA: usx2y: Annotate struct snd_usx2y_urb_seq with __counted_by
by Kees Cook 23 Sep '23

23 Sep '23

Prepare for the coming implementation by GCC and Clang of the __counted_by attribute. Flexible array members annotated with __counted_by can have their accesses bounds-checked at run-time checking via CONFIG_UBSAN_BOUNDS (for array indexing) and CONFIG_FORTIFY_SOURCE (for strcpy/memcpy-family functions). As found with Coccinelle[1], add __counted_by for struct snd_usx2y_urb_seq. Additionally, since the element count member must be set before accessing the annotated flexible array member, move its initialization earlier. [1] https://github.com/kees/kernel-tools/blob/trunk/coccinelle/examples/counted… Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/usb/usx2y/usbusx2y.h | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 8d82f5cc2fe1..391fd7b4ed5e 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -18,7 +18,7 @@ struct snd_usx2y_async_seq { struct snd_usx2y_urb_seq { int submitted; int len; - struct urb *urb[]; + struct urb *urb[] __counted_by(len); }; #include "usx2yhwdeppcm.h" diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 5197599e7aa6..ca7888495a9f 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -681,6 +681,7 @@ static int usx2y_rate_set(struct usx2ydev *usx2y, int rate) err = -ENOMEM; goto cleanup; } + us->len = NOOF_SETRATE_URBS; usbdata = kmalloc_array(NOOF_SETRATE_URBS, sizeof(int), GFP_KERNEL); if (!usbdata) { @@ -702,7 +703,6 @@ static int usx2y_rate_set(struct usx2ydev *usx2y, int rate) if (err < 0) goto cleanup; us->submitted = 0; - us->len = NOOF_SETRATE_URBS; usx2y->us04 = us; wait_event_timeout(usx2y->in04_wait_queue, !us->len, HZ); usx2y->us04 = NULL; -- 2.34.1

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[PATCH] ALSA: hda: Annotate struct hda_conn_list with __counted_by
by Kees Cook 23 Sep '23

23 Sep '23

Prepare for the coming implementation by GCC and Clang of the __counted_by attribute. Flexible array members annotated with __counted_by can have their accesses bounds-checked at run-time checking via CONFIG_UBSAN_BOUNDS (for array indexing) and CONFIG_FORTIFY_SOURCE (for strcpy/memcpy-family functions). As found with Coccinelle[1], add __counted_by for struct hda_conn_list. [1] https://github.com/kees/kernel-tools/blob/trunk/coccinelle/examples/counted… Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 33af707a65ab..01718b1fc9a7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -88,7 +88,7 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; - hda_nid_t conns[]; + hda_nid_t conns[] __counted_by(len); }; /* look up the cached results */ -- 2.34.1

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[PATCH] ASoC: loongson: Add error handling in loongson_pcm_open
by liuhaoran 23 Sep '23

23 Sep '23

This patch adds error-handling for the snd_pcm_hw_constraint_step() and snd_pcm_hw_constrain_integer() inside the img_prl_out_hw_params function in the loongson_dma.c file. Signed-off-by: liuhaoran <liuhaoran14(a)163.com> --- sound/soc/loongson/loongson_dma.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c index 47d6551903fc..af9e6a881a2a 100644 --- a/sound/soc/loongson/loongson_dma.c +++ b/sound/soc/loongson/loongson_dma.c @@ -240,17 +240,17 @@ static int loongson_pcm_open(struct snd_soc_component *component, ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); - if (ret<0) + if (ret < 0) return ret; - error = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128); - if (ret<0) + if (ret < 0) return ret; - error = snd_pcm_hw_constraint_integer(substream->runtime, + ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret<0) + if (ret < 0) return ret; snd_soc_set_runtime_hwparams(substream, &ls_pcm_hardware); -- 2.17.1

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[PATCH] ASoC: img: Add error handling for clk_set_rate in img_prl_out_hw_params
by liuhaoran 23 Sep '23

23 Sep '23

This patch adds error-handling for the clk_set_rate() call inside the img_prl_out_hw_params function in the img-parallel-out.c file. Signed-off-by: liuhaoran <liuhaoran14(a)163.com> --- sound/soc/img/img-parallel-out.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 815e68a7048c..698e9e92ded9 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -137,7 +137,10 @@ static int img_prl_out_hw_params(struct snd_pcm_substream *substream, if (channels != 2) return -EINVAL; - clk_set_rate(prl->clk_ref, rate * 256); + int ret = clk_set_rate(prl->clk_ref, rate * 256); + + if (!ret) + return ret; reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_PACKH_MASK) | control_set; -- 2.17.1

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[PATCH] ASoC: soc-generic-dmaengine-pcm: Fix function name in comment
by Miquel Raynal 22 Sep '23

22 Sep '23

While browsing/grepping in the sound core, I found that snd_dmaengine_set_config_from_dai_data() did not exist, in favor of snd_dmaengine_pcm_set_config_from_dai_data(). Fix the typo. Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d0653d775c87..cad222eb9a29 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, * platforms which make use of the snd_dmaengine_dai_dma_data struct for their * DAI DMA data. Internally the function will first call * snd_hwparams_to_dma_slave_config to fill in the slave config based on the - * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the - * remaining fields based on the DAI DMA data. + * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in + * the remaining fields based on the DAI DMA data. */ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) -- 2.34.1

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[PATCH v2] ASoC: dt-bindings: tfa9879: Convert to dtschema
by Bragatheswaran Manickavel 22 Sep '23

22 Sep '23

Convert the tfa9879 audio CODEC bindings to DT schema No error/warning seen when running make dt_binding_check Signed-off-by: Bragatheswaran Manickavel <bragathemanick0908(a)gmail.com> Changes: V1 -> V2: Fixed DT syntax errors and doc warning --- .../bindings/sound/nxp,tfa9879.yaml | 44 +++++++++++++++++++ .../devicetree/bindings/sound/tfa9879.txt | 23 ---------- MAINTAINERS | 2 +- 3 files changed, 45 insertions(+), 24 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tfa9879.txt diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml new file mode 100644 index 000000000000..df26248573ad --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa9879.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP TFA9879 class-D audio amplifier + +maintainers: + - Peter Rosin <peda(a)axentia.se> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nxp,tfa9879 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + i2c1 { + #address-cells = <1>; + #size-cells = <0>; + amplifier@6c { + compatible = "nxp,tfa9879"; + reg = <0x6c>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c1>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tfa9879.txt b/Documentation/devicetree/bindings/sound/tfa9879.txt deleted file mode 100644 index 1620e6848436..000000000000 --- a/Documentation/devicetree/bindings/sound/tfa9879.txt +++ /dev/null @@ -1,23 +0,0 @@ -NXP TFA9879 class-D audio amplifier - -Required properties: - -- compatible : "nxp,tfa9879" - -- reg : the I2C address of the device - -- #sound-dai-cells : must be 0. - -Example: - -&i2c1 { - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2c1>; - - amp: amp@6c { - #sound-dai-cells = <0>; - compatible = "nxp,tfa9879"; - reg = <0x6c>; - }; -}; - diff --git a/MAINTAINERS b/MAINTAINERS index a4c30221eb30..569303daf9b4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -15476,7 +15476,7 @@ NXP TFA9879 DRIVER M: Peter Rosin <peda(a)axentia.se> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/sound/tfa9879.txt +F: Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml F: sound/soc/codecs/tfa9879* NXP-NCI NFC DRIVER -- 2.34.1

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[PATCH v1 0/2] ALSA: cs35l41: prevent old firmwares using unsupported commands
by Stefan Binding 22 Sep '23

22 Sep '23

Some systems use older firmware which does not support newer commands which are used to enable external boost. For those systems, we can workaround this by writing the registers directly. We can use the firmware version, stored inside cs_dsp, to determine whether or not the command is supported. To achieve this, it requires a cleanup in the api, to pass the cs_dsp struct into the function. We can also remove the redundant boolean firmware_running from the HDA driver, and use the equivalent state inside cs_dsp. This chain is based on Mark's branch, since the api change was made to the function in sound/soc/codecs/cs35l41-lib.c. Stefan Binding (2): ALSA: hda: cs35l41: Remove unnecessary boolean state variable firmware_running ALSA: cs35l41: Fix for old systems which do not support command include/sound/cs35l41.h | 2 +- sound/pci/hda/cs35l41_hda.c | 28 ++++++++++++---------------- sound/pci/hda/cs35l41_hda.h | 1 - sound/soc/codecs/cs35l41-lib.c | 6 ++++-- sound/soc/codecs/cs35l41.c | 4 ++-- 5 files changed, 19 insertions(+), 22 deletions(-) -- 2.34.1

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