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[PATCH] ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
by Hui Wang 08 Jun '21

08 Jun '21

There are 2 issues on this machine, the 1st one is mic's plug/unplug can't be detected, that is because the mic is set to manual detecting mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto detecting mode. The other one is headphone's plug/unplug can't be detected by pulseaudio, that is because the pulseaudio will use ucm2/sof-hda-dsp on this machine, and the ucm2 only handle 'Headphone Jack', but on this machine the headphone's pincfg sets the location to Front, then the alsa mixer name is "Front Headphone Jack" instead of "Headphone Jack", so override the pincfg to change location to Left. BugLink: http://bugs.launchpad.net/bugs/1930188 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..be4a26f6bb96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,7 @@ enum { ALC285_FIXUP_HP_SPECTRE_X360, ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, + ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC283_FIXUP_HEADSET_MIC, }, + [ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x21, 0x03211030 }, /* Change the Headphone location to Left */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -8740,6 +8751,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, + {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

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[PATCH][next][V2] ASoC: rsnd: check for zero node count
by Colin King 08 Jun '21

08 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> Most callers of_get_child_count() check that "nr" is non-zero so it causes a static checker warning when we don't do that here. This does not cause a problem or a crash, but having zero SSUIes does not make sense either so let's add a check. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Rephrase commit message based on Dan Carpenter's suggestion. Thanks Dan! --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

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added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #95 was opened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 0/4] Support CS42L42 on JSL platform
by Brent Lu 07 Jun '21

07 Jun '21

This series consists four patches. Patch 1 adds dai sequence support for cml/jsl/tgl platforms which is different from the sequence on glk platform. Patch 2 adds max98360a support to the maxim-common module. Patch 3 adds driver data for jsl_cs4242_mx98360a which supports cs42l42 and max98360a running on jsl boards. Patch 4 refactor the sof_rt5682 to use the max98360 code in the maxim-common module. Brent Lu (4): ASoC: Intel: sof_cs42l42: support JSL DAI link sequence ASoC: Intel: maxim-common: support max98360a ASoC: intel: sof_cs42l42: add support for jsl_cs4242_mx98360a ASoC: Intel: sof_rt5682: code refactor for max98360a sound/soc/intel/boards/sof_cs42l42.c | 340 ++++++++++++------ sound/soc/intel/boards/sof_maxim_common.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 52 +-- .../intel/common/soc-acpi-intel-jsl-match.c | 8 + 5 files changed, 254 insertions(+), 167 deletions(-) -- 2.17.1

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[PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Geoffrey D. Bennett 07 Jun '21

07 Jun '21

These two patches add support for reading the mixer volumes and mux configuration from the hardware when the driver is initialising. Previously the ALSA volume controls were initialised to zero and the mux configuration set to a fixed default instead of being initialised to match the hardware state. The ALSA controls for the Scarlett Gen 2 interfaces should now always be in sync with the hardware. Thanks to Vladimir Sadovnikov for figuring out how to do this. Takashi, if these pass your review, I believe that they are appropriate for: #Cc: stable(a)vger.kernel.org Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Read mixer volumes at init time ALSA: usb-audio: scarlett2: Read mux at init time sound/usb/mixer_scarlett_gen2.c | 227 ++++++++++++++++++++++---------- 1 file changed, 161 insertions(+), 66 deletions(-) -- 2.31.1

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[PATCH][V2][next] ASoC: rk817: remove redundant assignment to pointer node, add missing of_node_put
by Colin King 07 Jun '21

07 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. The function is missing a of_node_put on node, fix this by adding the call before returning. Addresses-Coverity: ("Unused value") Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Add missing of_node_put call, kudos to Dan Carpenter for spotting this issue. --- sound/soc/codecs/rk817_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..a44d3cad1119 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -456,7 +456,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { @@ -466,6 +466,8 @@ static void rk817_codec_parse_dt_property(struct device *dev, rk817->mic_in_differential = of_property_read_bool(node, "rockchip,mic-in-differential"); + + of_node_put(node); } static int rk817_platform_probe(struct platform_device *pdev) -- 2.31.1

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[v7] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 07 Jun '21

07 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V6: -- Removed clk_disable OSR clock in lpass_cpu_daiops_prepare error case as failure of prepare will result in calling shutdown which should take care of this. Changes Since V5: -- Fixed nit. -- Updated `mi2s_was_prepared[dai->driver->id] = true;` after checking for errors. Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained. mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..647423a6cb57 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.32.0.rc1.229.g3e70b5a671-goog

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[PATCH v3 0/7] ASoC: adds new .auto_selectable_formats support
by Kuninori Morimoto 07 Jun '21

07 Jun '21

Hi Mark These are v3 of "ASoC: adds new .get_fmt support", but renamed Subject. This is a little bit challenging patch-set. The idea/code is almost same as v1 / v2. v3 has "priority" support. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .auto_selectable_formats. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had .auto_selectable_formats. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. v2 -> v3 - has priority - tidyup function explanation for snd_soc_dai_get_fmt() - Each driver don't try to have SND_SOC_DAIFMT_CBx_CFx to avoid confusion Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871racbx0w.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .auto_selectable_formats support ASoC: pcm3168a: add .auto_selectable_formats support ASoC: rsnd: add .auto_selectable_formats support ASoC: fsi: add .auto_selectable_formats support ASoC: hdmi-codec: add .auto_selectable_formats support include/sound/soc-dai.h | 55 +++++++ sound/soc/codecs/ak4613.c | 11 ++ sound/soc/codecs/hdmi-codec.c | 21 +++ sound/soc/codecs/pcm3168a.c | 26 +++ sound/soc/sh/fsi.c | 15 ++ sound/soc/sh/rcar/core.c | 31 +++- sound/soc/soc-core.c | 288 ++++++++++++++++++++++++++-------- sound/soc/soc-dai.c | 63 ++++++++ sound/soc/soc-utils.c | 29 ++++ 9 files changed, 475 insertions(+), 64 deletions(-) -- 2.25.1

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[PATCH] ASoC: rk817: Remove unneeded semicolon
by Jiapeng Chong 07 Jun '21

07 Jun '21

Fix the following coccicheck warnings: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rk817_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b..e95fe75 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,8 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } + return 0; } -- 1.8.3.1

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Re: [PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Takashi Iwai 07 Jun '21

07 Jun '21

On Mon, 07 Jun 2021 17:00:10 +0200, Vladimir Sadovnikov wrote: > > Hello! > > I would like to say some words from my side. > > The Scarlett device (especially 18i20) is pretty complicated device > and holds a lot of settings in it's internal > configuration area (hardware and software). > > So this is not the only patch which will configure the driver in proper way. > Since the device stores it's internal state (and that's good for power > safety and mobility), ideally, we should get > the almost fully compatible mixer settings with the original Focusrite > Control Software. > > The huge amount of job I've already done i my fork of Geoffrey's driver: > https://github.com/sadko4u/focusrite-scarlett-backports/blob/master/prod-dr… > > So we're planning to work on integrating our changes into the common > patch sets and will submit changes here. Sure, I don't mean against the patches, this looks like an acceptable approach. So don't worry, I'd take the patches once when the fixed version is submitted. However, from the system design POV, all those configurations should be a software issue, and ideally we shouldn't rely on the hardware preset state which has been done *somehow* -- it may allow malfunction easily. One thing I've learned over years is that you can never trust hardware :) Takashi > > Best, > Vladimir > > 07.06.2021 10:23, Takashi Iwai пишет: > > On Sun, 06 Jun 2021 16:16:44 +0200, > > Geoffrey D. Bennett wrote: > >> These two patches add support for reading the mixer volumes and mux > >> configuration from the hardware when the driver is initialising. > >> > >> Previously the ALSA volume controls were initialised to zero and the > >> mux configuration set to a fixed default instead of being initialised > >> to match the hardware state. > >> > >> The ALSA controls for the Scarlett Gen 2 interfaces should now always > >> be in sync with the hardware. Thanks to Vladimir Sadovnikov for > >> figuring out how to do this. > >> > >> Takashi, if these pass your review, I believe that they are > >> appropriate for: > >> #Cc: stable(a)vger.kernel.org > > Well, in general, having a proper fixed value for the initial mixer > > value is the right thing, which is a part of the driver's role. > > Though, in snd-usb-audio, we don't set up the initial values just > > because of laziness; since the topology in USB audio is variable per > > device and often hard to parse correctly, it's difficult to determine > > the suitable initial values, hence we leave untouched. So, in that > > sense, setting the zero isn't wrong, rather safer, per se. > > > > However, Scarlett 2 seems to want to be different; it has already some > > initialization code to read the existing configs. So this change > > sounds more or less acceptable. But it's questionable whether it's > > really for stable as a "fix". > > > > In anyway, please fix the bug ktest bot spotted, the missing endian > > conversions and resubmit. > > > > > > thanks, > > > > Takashi > >

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