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

[PATCH v2 1/2] ASoC: cs35l41: Add one more variable in the debug log
by Hui Wang 28 Mar '22

28 Mar '22

otp_map[].size is a key variable to compute the value of otp_val and to update the bit_offset, it is helpful to debug if could put it in the debug log. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/cs35l41-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index e5a56bcbb223..d0a480c40231 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -822,8 +822,8 @@ int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap) word_offset = otp_map_match->word_offset; for (i = 0; i < otp_map_match->num_elements; i++) { - dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d\n", - bit_offset, word_offset, bit_sum % 32); + dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d otp_map[i].size = %d\n", + bit_offset, word_offset, bit_sum % 32, otp_map[i].size); if (bit_offset + otp_map[i].size - 1 >= 32) { otp_val = (otp_mem[word_offset] & GENMASK(31, bit_offset)) >> bit_offset; -- 2.25.1

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[PATCH v1 0/3] Exynos Usb Audio Offloading Support
by Oh Eomji 28 Mar '22

28 Mar '22

Exynos uses the usb audio offloading functions to save power consumption in the usb audio device. The audio control interface is processed in the existing usb path, and the audio stream interface is processed in the audio path. Through communication between AP usb and audio usb f/w, usb audio device connection information, xhci memory information, etc. are notified to usb audio f/w so that the abox directly controls xhci transmission. Vendor's hooking interface is required for this functions. Throught this interface, information such as usb audio device connection information, pcm interface information, sample rate setting, xhci memory, full descriptor of connected device, and setting point can be transmitted to usb audio f/w. the usb audio f/w can set up and interface, transmit audio data, etc. through the received information. This patchset includes the following for using usb audio offloading: - vendor's hooking interface - vendor's hooking interface calling point - user using vendor's hooking interface Oh Eomji (3): sound: usb: Add vendor's hooking interface sound: usb: Calling vendor's call-back function within usb audio operation. sound: usb: Exynos usb audio offloading driver sound/usb/Kconfig | 9 + sound/usb/Makefile | 2 + sound/usb/card.c | 119 +++++++++ sound/usb/card.h | 20 ++ sound/usb/exynos_usb_audio.c | 560 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/exynos_usb_audio.h | 150 ++++++++++++ sound/usb/pcm.c | 37 +++ sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 45 ++++ 9 files changed, 944 insertions(+) create mode 100644 sound/usb/exynos_usb_audio.c create mode 100644 sound/usb/exynos_usb_audio.h -- 2.7.4

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[PATCH] soc: soc-core: fix a missing check on list iterator
by Xiaomeng Tong 28 Mar '22

28 Mar '22

The bug is here: *dai_name = dai->driver->name; For for_each_component_dais, just like list_for_each_entry, the list iterator 'dai' will point to a bogus position containing HEAD if the list is empty or no element is found. This case must be checked before any use of the iterator, otherwise it will lead to a invalid memory access. To fix the bug, use a new variable 'iter' as the list iterator, while use the original variable 'dai' as a dedicated pointer to point to the found element. Cc: stable(a)vger.kernel.org Fixes: 58bf4179000a3 ("ASoC: soc-core: remove dai_drv from snd_soc_component") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/soc/soc-core.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 434e61b46983..064fc0347868 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3238,7 +3238,7 @@ int snd_soc_get_dai_name(const struct of_phandle_args *args, ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name); if (ret == -ENOTSUPP) { - struct snd_soc_dai *dai; + struct snd_soc_dai *dai = NULL, *iter; int id = -1; switch (args->args_count) { @@ -3261,12 +3261,19 @@ int snd_soc_get_dai_name(const struct of_phandle_args *args, ret = 0; /* find target DAI */ - for_each_component_dais(pos, dai) { - if (id == 0) + for_each_component_dais(pos, iter) { + if (id == 0) { + dai = iter; break; + } id--; } + if (!dai) { + ret = -EINVAL; + continue; + } + *dai_name = dai->driver->name; if (!*dai_name) *dai_name = pos->name; -- 2.17.1

2 1

Audio only working after stopping PulseAudio and setting the UCM - kblda7219max on the ChromeOS Kernel, 5.10.70
by GitHub issues - edited 28 Mar '22

28 Mar '22

alsa-project/alsa-ucm-conf issue #142 was edited from MilkyDeveloper: On my `kblda7219max` Chromebook (baseboard: NAMI), internal audio works properly when the following conditions are met: * Kernel 5.10.70 (ChromeOS fork) is used (the latest ChromeOS kernel has some kind of topology-related kernel bug) * FW is copied from the ChromeOS rootfs to Linux * UCM is converted to UCM2 * Only ALSA is used However, whenever I start PulseAudio (`pulseaudio -vvvv`), audio is disabled, going to a dummy output. PulseAudio passes through probing for the UCM files but errors out in this particular spot: ``` I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying hw:kblda7219max,0 without SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:kblda7219max,0 D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 21845 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) ``` which causes it to say: ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-kbl_da7219_mx98357a" card_name="alsa_card.platform-kbl_da7219_mx98357a" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` My `dmesg` gets filled with: ``` Kbl Audio Port: ASoC: no backend DAIs enabled for Kbl Audio Port Kbl Audio Port: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` However, once I kill PulseAudio and run `alsaucm -c kblda7219max set _verb HiFi set _enadev Speaker`, audio begins working again (but this is in pure ALSA and PA apps are not compatible) and the `no backend DAIs enabled for Kbl Audio Port` is no longer logged to `dmesg`. I think this is a UCM issue because the audio is working fine kernel-level and with ALSA, but PulseAudio seems to mess something up with the UCM and the UCM needs to be set again. Huge thanks for any help! [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355612/… (My PulseAudio log is slightly different from the time I ran PulseAudio and got the PCM_IOCTL_PREPARE error) [pulseaudio-log.txt](https://github.com/alsa-project/alsa-ucm-conf/files/835… [HiFi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355631/… [kblda7219max.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/142 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] soc: meson: fix a missing check on list iterator
by Xiaomeng Tong 27 Mar '22

27 Mar '22

The bug is here: *dai_name = dai->driver->name; For for_each_component_dais(), just like list_for_each_entry, the list iterator 'runtime' will point to a bogus position containing HEAD if the list is empty or no element is found. This case must be checked before any use of the iterator, otherwise it will lead to a invalid memory access. To fix the bug, just move the assignment into loop and return 0 when element is found, otherwise return -EINVAL; Cc: stable(a)vger.kernel.org Fixes: 6ae9ca9ce986b ("ASoC: meson: aiu: add i2s and spdif support") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/soc/meson/aiu.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index d299a70db7e5..b52915c6f53b 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -61,14 +61,14 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return -EINVAL; for_each_component_dais(component, dai) { - if (id == 0) - break; + if (id == 0) { + *dai_name = dai->driver->name; + return 0; + } id--; } - *dai_name = dai->driver->name; - - return 0; + return -EINVAL; } static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, -- 2.17.1

2 3

[PATCH] cs423x: cs4236: fix an incorrect NULL check on list iterator
by Xiaomeng Tong 27 Mar '22

27 Mar '22

The bug is here: err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); The list iterator value 'cdev' will *always* be set and non-NULL by list_for_each_entry(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found. To fix the bug, use a new variable 'iter' as the list iterator, while use the original variable 'cdev' as a dedicated pointer to point to the found element. And snd_card_cs423x_pnp() itself has NULL check for cdev. Cc: stable(a)vger.kernel.org Fixes: c2b73d1458014 ("ALSA: cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- sound/isa/cs423x/cs4236.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index b6bdebd9ef27..10112e1bb25d 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -494,7 +494,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static int dev; int err; struct snd_card *card; - struct pnp_dev *cdev; + struct pnp_dev *cdev, *iter; char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) @@ -510,9 +510,11 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, strcpy(cid, pdev->id[0].id); cid[5] = '1'; cdev = NULL; - list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { - if (!strcmp(cdev->id[0].id, cid)) + list_for_each_entry(iter, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(iter->id[0].id, cid)) { + cdev = iter; break; + } } err = snd_cs423x_card_new(&pdev->dev, dev, &card); if (err < 0) -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek: Enable headset mic on Lenovo P360
by Kai-Heng Feng 27 Mar '22

27 Mar '22

Lenovo P360 is another platform equipped with ALC897, and it needs ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e88fbef57c40c..4c33cb57963db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11152,6 +11152,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), -- 2.34.1

2 1

PulseAudio only working after setting the UCM - kblda7219max on the ChromeOS Kernel, 5.10.70
by GitHub issues - opened 26 Mar '22

26 Mar '22

alsa-project/alsa-ucm-conf issue #142 was opened from MilkyDeveloper: On my `kblda7219max` Chromebook (baseboard: NAMI), internal audio works properly when the following conditions are met: * Kernel 5.10.70 (ChromeOS fork) is used (the latest ChromeOS kernel has some kind of topology-related kernel bug) * FW is copied from the ChromeOS rootfs to Linux * UCM is converted to UCM2 * Only ALSA is used However, whenever I start PulseAudio (`pulseaudio -vvvv`), audio is disabled, going to a dummy output. PulseAudio passes through probing for the UCM files but errors out in this particular spot: ``` I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying hw:kblda7219max,0 without SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:kblda7219max,0 D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 21845 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) ``` which causes it to say: ``` E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-kbl_da7219_mx98357a" card_name="alsa_card.platform-kbl_da7219_mx98357a" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` My `dmesg` gets filled with: ``` Kbl Audio Port: ASoC: no backend DAIs enabled for Kbl Audio Port Kbl Audio Port: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` However, once I kill PulseAudio and run `alsaucm -c kblda7219max set _verb HiFi set _enadev Speaker`, audio begins working again (but this is in pure ALSA and PA apps are not compatible) and the `no backend DAIs enabled for Kbl Audio Port` is no longer logged to `dmesg`. I think this is a UCM issue because the audio is working fine kernel-level and with ALSA, but PulseAudio seems to mess something up with the UCM and the UCM needs to be set again. Huge thanks for any help! [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355612/… (My PulseAudio log is slightly different from the time I ran PulseAudio and got the PCM_IOCTL_PREPARE error) [pulseaudio-log.txt](https://github.com/alsa-project/alsa-ucm-conf/files/835… [HiFi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8355631/… [kblda7219max.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/142 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Conceptual bug on SoundWire probe/remove?
by Pierre-Louis Bossart 26 Mar '22

26 Mar '22

Hi, I could use feedback/guidance on a possible conceptual bug in the SoundWire probe and bus handling. When we probe a driver, the code does this: static int sdw_drv_probe(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); struct sdw_driver *drv = drv_to_sdw_driver(dev->driver); const struct sdw_device_id *id; const char *name; int ret; /* * fw description is mandatory to bind */ if (!dev->fwnode) return -ENODEV; if (!IS_ENABLED(CONFIG_ACPI) && !dev->of_node) return -ENODEV; id = sdw_get_device_id(slave, drv); if (!id) return -ENODEV; slave->ops = drv->ops; The last line is the problematic one. If at some point, the user does an rmmod and unbinds the SoundWire codec driver, the .remove will be called and the 'drv' will no longer be valid, but we will still have a reference to drv->ops and use that pointer in the bus code, e.g. /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); slave->ops->interrupt_callback(slave, &slave_intr); } I noodled with a potential fix in https://github.com/thesofproject/linux/pull/3534/commits/82d64fb0fd39b53226… where I force-reset this slave->ops pointer, but it is likely to be very racy. We probably need to avoid such references, or have a clean mechanism to unbind, e.g. with all commands and interrupts stopped while the codec driver .remove routine is handled. Initial error reports at https://github.com/thesofproject/linux/issues/3531 Suggestions and comments welcome, thanks! -Pierre

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[PATCH] ASoC: codecs: add support for ES8326 dts
by zhuning＠everest-semi.com 26 Mar '22

26 Mar '22

From: yangxiaohua <yangxiaohua(a)everest-semi.com> Subject: [PATCH 1/1] ASoC: codecs: add support for ES8326 dts The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. This is a separate patch for ES8326 dts Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> --- .../bindings/sound/everest,es8326.yaml | 109 ++++++++++++++++++ 1 file changed, 109 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..8cae9c328dbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + mclk-rate: + description: + clock rate of mclk + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + + everest,jack-pol: + description: + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the bottom on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + description: + the value of reg 2A when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + description: + the value of reg 2A when headset unplugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,amic1-src: + description: + the value of reg 23 when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,amic2-src: + description: + the value of reg 23 when headset unplugged. + Set to 0x19 if dmic used. Set to 0x29 if amic used. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,jack-detect-inverted: + description: + Defined to invert the jack detection. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + - "#sound-dai-cells" + + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + mic1-src = [22]; + mic2-src = [44]; + amic1-src = [19]; + amic2-src = [19]; + jack-pol = [0e]; + }; + }; -- 2.17.1

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