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

[PATCH v1 00/11] Fix support for System Suspend for CS35L41 HDA
by Stefan Binding 21 Jul '23

21 Jul '23

There have been a couple of customer reports of intermittant issues after system resume, where sometimes the DSP firmware stops responding. Investigations into this issue show that there is a race between receiving a prepare from the HDA core, and the firmware reload which is started by the system resume. This can causes the Global Enable on the CS35L41 to be enabled during the firmware load, which can sometimes cause issues in the DSP. The existing system resume behaviour also did not resume the audio, if audio was previously playing when it was suspended. In addition, during investigation, it was found there were additional problems in the System Resume sequence, as well as the Playback sequence with External Boost, where the driver does not correctly follow its enable sequence for this mode. This can cause additional issues such as pops and clicks. This chain intends to correct the sequences for playback and system suspend/resume so that the driver: obeys the external boost enable sequence; resumes audio on system resume; and avoids the race condition on firmware load and playback during system resume. Stefan Binding (11): ALSA: cs35l41: Use mbox command to enable speaker output for external boost ALSA: hda: cs35l41: Check mailbox status of pause command after firmware load ALSA: hda: cs35l41: Ensure we correctly re-sync regmap before system suspending. ALSA: hda: cs35l41: Ensure we pass up any errors during system suspend. ALSA: hda: cs35l41: Move Play and Pause into separate functions ALSA: hda: hda_component: Add pre and post playback hooks to hda_component ALSA: hda: cs35l41: Use pre and post playback hooks ALSA: hda: cs35l41: Rework System Suspend to ensure correct call separation ALSA: hda/realtek: Support pre-/post- playback hooks for cs35l41 ALSA: hda: cs35l41: Add device_link between HDA and cs35l41_hda ALSA: hda: cs35l41: Ensure amp is only unmuted during playback include/sound/cs35l41.h | 5 +- sound/pci/hda/cs35l41_hda.c | 288 +++++++++++++++++++++++++-------- sound/pci/hda/hda_component.h | 2 + sound/pci/hda/patch_realtek.c | 10 +- sound/soc/codecs/cs35l41-lib.c | 118 ++++++++++++-- sound/soc/codecs/cs35l41.c | 18 +-- 6 files changed, 343 insertions(+), 98 deletions(-) -- 2.34.1

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[PATCH 0/2] ALSA: hda/hdmi: Add Intel Lunar Lake support
by Peter Ujfalusi 21 Jul '23

21 Jul '23

Hi, after fixing the ordering, add the HDMI codec ID for LNL Regards, Peter --- Kai Vehmanen (2): ALSA: hda/hdmi: keep codec entries in numerical order ALSA: hda: add HDMI codec ID for Intel LNL sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 2.41.0

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[PATCH v1 1/1] ALSA: korg1212: Re-use sockptr_t and respective APIs
by Andy Shevchenko 21 Jul '23

21 Jul '23

The sockptr_t (despite the naming) is a generic type to hold kernel or user pointer and there are respective APIs to copy data to or from it. Replace open coded variants in the driver by them. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/korg1212/korg1212.c | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 33b4f95d65b3..92c3eab4d12c 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -10,6 +10,7 @@ #include <linux/interrupt.h> #include <linux/pci.h> #include <linux/slab.h> +#include <linux/sockptr.h> #include <linux/wait.h> #include <linux/module.h> #include <linux/mutex.h> @@ -1285,8 +1286,7 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun } static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, - void __user *dst, int pos, int count, - bool in_kernel) + sockptr_t dst, int pos, int count) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); @@ -1306,24 +1306,21 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + printk(KERN_DEBUG "K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, src, sockptr_is_kernel(dst) ? dst.kernel : dst.user, i); return -EFAULT; } #endif - if (in_kernel) - memcpy((__force void *)dst, src, size); - else if (copy_to_user(dst, src, size)) + if (copy_to_sockptr_offset(dst, i * size, src, size)) return -EFAULT; src++; - dst += size; } return 0; } static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, - void __user *src, int pos, int count, - bool in_kernel) + sockptr_t src, int pos, int count) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); @@ -1345,16 +1342,14 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + printk(KERN_DEBUG "K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, sockptr_is_kernel(src) ? src.kernel : src.user, dst, i); return -EFAULT; } #endif - if (in_kernel) - memcpy(dst, (__force void *)src, size); - else if (copy_from_user(dst, src, size)) + if (copy_from_sockptr_offset(dst, src, i * size, size)) return -EFAULT; dst++; - src += size; } return 0; @@ -1644,15 +1639,14 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, void __user *src, unsigned long count) { - return snd_korg1212_copy_from(substream, src, pos, count, false); + return snd_korg1212_copy_from(substream, USER_SOCKPTR(src), pos, count); } static int snd_korg1212_playback_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, void *src, unsigned long count) { - return snd_korg1212_copy_from(substream, (void __user *)src, - pos, count, true); + return snd_korg1212_copy_from(substream, KERNEL_SOCKPTR(src), pos, count); } static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, @@ -1672,15 +1666,14 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, void __user *dst, unsigned long count) { - return snd_korg1212_copy_to(substream, dst, pos, count, false); + return snd_korg1212_copy_to(substream, USER_SOCKPTR(dst), pos, count); } static int snd_korg1212_capture_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, void *dst, unsigned long count) { - return snd_korg1212_copy_to(substream, (void __user *)dst, - pos, count, true); + return snd_korg1212_copy_to(substream, KERNEL_SOCKPTR(dst), pos, count); } static const struct snd_pcm_ops snd_korg1212_playback_ops = { -- 2.40.0.1.gaa8946217a0b

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[PATCH v2] ASoC: wm8960: Add DAC filter characteristics selection
by Shengjiu Wang 21 Jul '23

21 Jul '23

Support DAC filter characteristics selection: Normal mode and Sloping stopband. Sloping stopband may have better frequency response. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- changes in v2: - modify capitalise sound/soc/codecs/wm8960.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 366f5d769d6d..5db9b0338396 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -155,6 +155,7 @@ static const char *wm8960_adc_data_output_sel[] = { "Left Data = Right ADC; Right Data = Left ADC", }; static const char *wm8960_dmonomix[] = {"Stereo", "Mono"}; +static const char *wm8960_dacslope[] = {"Normal", "Sloping"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -165,6 +166,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 4, 2, wm8960_dmonomix), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 1, 2, wm8960_dacslope), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -307,6 +309,7 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), SOC_ENUM("DAC Mono Mix", wm8960_enum[7]), +SOC_ENUM("DAC Filter Characteristics", wm8960_enum[8]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 2.34.1

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[PATCH -next] ALSA: seq: remove redundant unsigned comparison to zero
by Wang Weiyang 21 Jul '23

21 Jul '23

Since struct snd_ump_block_info::first_group is unsigned char, comparison to zero is redundant Signed-off-by: Wang Weiyang <wangweiyang2(a)huawei.com> --- sound/core/seq/seq_ump_client.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index fe21c801af74..f26a1812dfa7 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -298,8 +298,7 @@ static void update_group_attrs(struct seq_ump_client *client) } list_for_each_entry(fb, &client->ump->block_list, list) { - if (fb->info.first_group < 0 || - fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) + if (fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) break; group = &client->groups[fb->info.first_group]; for (i = 0; i < fb->info.num_groups; i++, group++) { -- 2.25.1

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[PATCH v1] ASoC: dt-bindings: wm8904: Convert to dtschema
by Francesco Dolcini 21 Jul '23

21 Jul '23

From: Francesco Dolcini <francesco.dolcini(a)toradex.com> Convert the WM8904 audio CODEC bindings to DT schema. Compared to the original binding #sound-dai-cells and the missing power supplies are added. The latter are all required as described in the datasheet. Datasheet: https://statics.cirrus.com/pubs/proDatasheet/WM8904_Rev4.1.pdf Signed-off-by: Francesco Dolcini <francesco.dolcini(a)toradex.com> --- .../devicetree/bindings/sound/wlf,wm8904.yaml | 74 +++++++++++++++++++ .../devicetree/bindings/sound/wm8904.txt | 33 --------- 2 files changed, 74 insertions(+), 33 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8904.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000000..329260cf0fa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8904/WM8912 audio codecs + +maintainers: + - patches(a)opensource.cirrus.com + +description: | + Pins on the device (for linking into audio routes): + IN1L, IN1R, IN2L, IN2R, IN3L, IN3R, HPOUTL, HPOUTR, LINEOUTL, LINEOUTR, + MICBIAS + +properties: + compatible: + enum: + - wlf,wm8904 + - wlf,wm8912 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + AVDD-supply: true + CPVDD-supply: true + DBVDD-supply: true + DCVDD-supply: true + MICVDD-supply: true + +required: + - compatible + - reg + - clocks + - clock-names + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; + AVDD-supply = <&reg_1p8v>; + CPVDD-supply = <&reg_1p8v>; + DBVDD-supply = <&reg_1p8v>; + DCVDD-supply = <&reg_1p8v>; + MICVDD-supply = <&reg_1p8v>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt deleted file mode 100644 index 66bf261423b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8904.txt +++ /dev/null @@ -1,33 +0,0 @@ -WM8904 audio CODEC - -This device supports I2C only. - -Required properties: - - compatible: "wlf,wm8904" or "wlf,wm8912" - - reg: the I2C address of the device. - - clock-names: "mclk" - - clocks: reference to - <Documentation/devicetree/bindings/clock/clock-bindings.txt> - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - -Examples: - -codec: wm8904@1a { - compatible = "wlf,wm8904"; - reg = <0x1a>; - clocks = <&pck0>; - clock-names = "mclk"; -}; -- 2.25.1

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[PATCH v1 0/2] ASoC: dt-bindings: nau8822: minor updates
by Francesco Dolcini 21 Jul '23

21 Jul '23

From: Francesco Dolcini <francesco.dolcini(a)toradex.com> Minor updates to NAU8822 DT bindings: - Add #sound-dai-cells - Add MCLK Francesco Dolcini (2): ASoC: dt-bindings: nau8822: Add #sound-dai-cells ASoC: dt-bindings: nau8822: Add MCLK clock .../devicetree/bindings/sound/nuvoton,nau8822.yaml | 12 ++++++++++++ 1 file changed, 12 insertions(+) -- 2.25.1

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Fwd: 6.4 and higher causes audio distortion
by Bagas Sanjaya 21 Jul '23

21 Jul '23

Hi, I notice a regression report on Bugzilla [1]. Quoting from it: > I have a Lenovo ThinkPad X1 Yoga Gen 7 running Arch Linux. Linux 6.4 and higher, cause audio distortion. Sometimes, this occurs to the point that nearly nothing is discernible. This carries over to wired headphones. The issue occurs on the entire mainline 6.4.x kernel series and also the 6.4.3 stable and 6.5 RC1 kernel, which are the latest at the time of writing. The issue occurs on both the Arch distributed kernels, and the mainline kernels. > > Linux kernels 6.3.x are not affected and neither is the 6.1 LTS kernel series which is what I am temporarily using. On Windows 10/11 too, the audio works as it should. This indicates that my hardware is not at fault. Bluetooth audio is not impacted from my testing, either. > > The distortion doesn't start immediately. It either occurs automatically after a random amount of time, or when I increase/decrease the volume, or when I skip forward/backward to a section. In order to stop the distortion, I have to either increase/decrease the volume until it stops, or skip forward/backward until it stops, or restart Pipewire via systemd, however it starts again due to one of the aforementioned reasons. > > At the time of this report, I am running Pipewire 0.3.74 and Wireplumber 0.4.14. This also doesn't seem like a Pipewire/Wireplumber issue, since these same versions work fine on the 6.1 LTS kernels without causing any audio distortion. > > I wrote about this on the Arch Linux forums, too, and seems like at least two other people are facing this issue. Here's the forum post: https://bbs.archlinux.org/viewtopic.php?id=287068 > > Furthermore, I filed a bug report on the Arch Linux Bug Reporter, where they suggested that the issue is a kernel regression and should be reported upstream, here. Here's the bug report that I filed on the Arch Linux Bug Reporter for anyone interested: https://bugs.archlinux.org/task/79081?project=1&pagenum=10 > > I have attached the dmesg outputs of the mainline 6.5 RC1 kernel. > > Here's some audio related hardware information from my device: > > inxi -A > > Audio: > Device-1: Intel Alder Lake PCH-P High Definition Audio > driver: sof-audio-pci-intel-tgl > API: ALSA v: k6.5.0-rc1-1-mainline status: kernel-api > > > pactl info > > Server String: /run/user/1000/pulse/native > Library Protocol Version: 35 > Server Protocol Version: 35 > Is Local: yes > Client Index: 138 > Tile Size: 65472 > User Name: tux > Host Name: NSA-Terminal-4 > Server Name: PulseAudio (on PipeWire 0.3.74) > Server Version: 15.0.0 > Default Sample Specification: float32le 2ch 48000Hz > Default Channel Map: front-left,front-right > Default Sink: alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__hw_sofhdadsp__sink > Default Source: alsa_input.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__hw_sofhdadsp_6__source > Cookie: f9dc:5e7a > > > I can't figure out why this is happening. Kindly ask for any more information that is necessary. Thank you. See Bugzilla for the full thread and attached dmesg. Anyway, I'm adding it to regzbot so that it doesn't fall through cracks unnoticed: #regzbot introduced: v6.3..v6.4 https://bugzilla.kernel.org/show_bug.cgi?id=217673 #regzbot title: PipeWire + Wireplumber audio distortion on Lenovo ThinkPad X1 Yoga Gen 7 #regzbot link: https://bbs.archlinux.org/viewtopic.php?id=287068 #regzbot link: https://bugs.archlinux.org/task/79081?project=1&pagenum=10 Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=217673 -- An old man doll... just what I always wanted! - Clara

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[PATCH v2 0/5] fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> Currently, the codec driver could re-enables SDCA interrupt mask and the bus driver re-enables Imp-defined interrupt mask when the devices are re-attached. If the manager uses ClockStop Mode0 to peripherals, the SDCA/Imp-defined interrupt mask will not be enabled when the system resumes. These patches will fix this issue. -- v2 - modify the commit messages Shuming Fan (5): ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0 ASoC: rt711: fix for JD event handling in ClockStop Mode0 ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0 sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 5 files changed, 43 insertions(+), 5 deletions(-) -- 2.34.1

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[PATCH] ASoC: wm8960: Add DAC filter characteristics selection
by Shengjiu Wang 21 Jul '23

21 Jul '23

Support DAC filter characteristics selection: Normal mode and Sloping stopband. Sloping stopband may have better frequency response. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 366f5d769d6d..ff1355306b43 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -155,6 +155,7 @@ static const char *wm8960_adc_data_output_sel[] = { "Left Data = Right ADC; Right Data = Left ADC", }; static const char *wm8960_dmonomix[] = {"Stereo", "Mono"}; +static const char *wm8960_dacslope[] = {"Normal", "Sloping"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -165,6 +166,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 4, 2, wm8960_dmonomix), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 1, 2, wm8960_dacslope), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -307,6 +309,7 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), SOC_ENUM("DAC Mono Mix", wm8960_enum[7]), +SOC_ENUM("DAC filter characteristics", wm8960_enum[8]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 2.34.1

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