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

[PATCH AUTOSEL 5.19 13/22] ALSA: hda/tegra: Align BDL entry to 4KB boundary
by Sasha Levin 14 Sep '22

14 Sep '22

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 8d44e6044a0e885acdd01813768a0b27906d64fd ] AZA HW may send a burst read/write request crossing 4K memory boundary. The 4KB boundary is not guaranteed by Tegra HDA HW. Make SW change to include the flag AZX_DCAPS_4K_BDLE_BOUNDARY to align BDLE to 4K boundary. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Link: https://lore.kernel.org/r/20220905172420.3801-1-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7debb2c76aa62..976a112c7d006 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -474,7 +474,8 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { const unsigned int driver_flags = AZX_DCAPS_CORBRP_SELF_CLEAR | - AZX_DCAPS_PM_RUNTIME; + AZX_DCAPS_PM_RUNTIME | + AZX_DCAPS_4K_BDLE_BOUNDARY; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; -- 2.35.1

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[PATCH AUTOSEL 5.19 11/22] ALSA: hda/sigmatel: Keep power up while beep is enabled
by Sasha Levin 14 Sep '22

14 Sep '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 414d38ba871092aeac4ed097ac4ced89486646f7 ] It seems that the beep playback doesn't work well on IDT codec devices when the codec auto-pm is enabled. Keep the power on while the beep switch is enabled. Link: https://bugzilla.suse.com/show_bug.cgi?id=1200544 Link: https://lore.kernel.org/r/20220904072750.26164-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_sigmatel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61df4d33c48ff..066bfccd25877 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -209,6 +209,7 @@ struct sigmatel_spec { /* beep widgets */ hda_nid_t anabeep_nid; + bool beep_power_on; /* SPDIF-out mux */ const char * const *spdif_labels; @@ -4443,6 +4444,26 @@ static int stac_suspend(struct hda_codec *codec) return 0; } + +static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + int ret = snd_hda_gen_check_power_status(codec, nid); + +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (nid == spec->gen.beep_nid && codec->beep) { + if (codec->beep->enabled != spec->beep_power_on) { + spec->beep_power_on = codec->beep->enabled; + if (spec->beep_power_on) + snd_hda_power_up_pm(codec); + else + snd_hda_power_down_pm(codec); + } + ret |= spec->beep_power_on; + } +#endif + return ret; +} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4455,6 +4476,7 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, + .check_power_status = stac_check_power_status, #endif }; -- 2.35.1

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[PATCH AUTOSEL 5.19 06/22] ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable
by Sasha Levin 14 Sep '22

14 Sep '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit b1cd3fd42db7593a2d24c06f1c53b8c886592080 ] Call pm_runtime_disable() when error happens in probe() Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1661430460-5234-2-git-send-email-shengjiu.wang@nx… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index e09015e7e3c7c..38f3863739f4d 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -240,6 +240,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "failed to pcm register\n"); + pm_runtime_disable(&pdev->dev); return ret; } @@ -248,6 +249,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) &fsl_aud2htx_dai, 1); if (ret) { dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + pm_runtime_disable(&pdev->dev); return ret; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 05/22] ASoC: fsl_aud2htx: register platform component before registering cpu dai
by Sasha Levin 14 Sep '22

14 Sep '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit ea532c29972df96fda20393d9bf057e898f5e965 ] There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. And the behavior of imx_pcm_dma_init() is same as common devm_snd_dmaengine_pcm_register(), so use devm_snd_dmaengine_pcm_register() instead Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1661430460-5234-1-git-send-email-shengjiu.wang@nx… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 422922146f2a5..e09015e7e3c7c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -233,6 +233,16 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) regcache_cache_only(aud2htx->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_aud2htx_component, &fsl_aud2htx_dai, 1); @@ -241,10 +251,6 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); - if (ret) - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - return ret; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 03/22] ASoC: nau8824: Fix semaphore unbalance at error paths
by Sasha Levin 14 Sep '22

14 Sep '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5628560e90395d3812800a8e44a01c32ffa429ec ] The semaphore of nau8824 wasn't properly unlocked at some error handling code paths, hence this may result in the unbalance (and potential lock-up). Fix them to handle the semaphore up properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20220823081000.2965-3-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 2a7c935085353..c263858745ee2 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1043,6 +1043,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + int err = -EINVAL; nau8824_sema_acquire(nau8824, HZ); @@ -1059,7 +1060,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_DAC_OVERSAMPLE_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); @@ -1069,7 +1070,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_ADC_SYNC_DOWN_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_ADC_SRC_MASK, osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); @@ -1090,7 +1091,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, else if (bclk_fs <= 256) bclk_div = 0; else - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_2, NAU8824_I2S_LRC_DIV_MASK | NAU8824_I2S_BLK_DIV_MASK, @@ -1111,15 +1112,17 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8824_I2S_DL_32; break; default: - return -EINVAL; + goto error; } regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, val_len); + err = 0; + error: nau8824_sema_release(nau8824); - return 0; + return err; } static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -1128,8 +1131,6 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8824_sema_acquire(nau8824, HZ); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8824_I2S_MS_MASTER; @@ -1171,6 +1172,8 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DF_MASK | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_EN, ctrl1_val); -- 2.35.1

1 0

[PATCH linux-next] soundwire: intel: Remove unused parameter ret
by cgel.zte＠gmail.com 14 Sep '22

14 Sep '22

From: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> The parameter ret = 0 has not been used. Return 0 directly instead of returning ret. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> --- drivers/soundwire/intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index b006ae6efcb0..26fdf3577454 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -395,7 +395,6 @@ static int intel_shim_init(struct sdw_intel *sdw, bool clock_stop) { void __iomem *shim = sdw->link_res->shim; unsigned int link_id = sdw->instance; - int ret = 0; u16 ioctl = 0, act = 0; mutex_lock(sdw->link_res->shim_lock); @@ -427,7 +426,7 @@ static int intel_shim_init(struct sdw_intel *sdw, bool clock_stop) mutex_unlock(sdw->link_res->shim_lock); - return ret; + return 0; } static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) -- 2.25.1

2 1

[PATCH] ALSA: patch_realtek: Fix Dell Inspiron Plus 16
by Philipp Jungkamp 14 Sep '22

14 Sep '22

The Dell Inspiron Plus 16 has top speakers connected on NID 0x17, which the codec reports as unconnected. These speakers should be connected to the DAC on NID 0x03. --- I don't possess this Laptop myself. The owner of this Laptop reports that the speakers are now active, though there are still problem's. > There are still 2 small issues present: > 1. 0,5s delay before the top speakers turn on > 2. Popping sound from headphones 5s after the music stops playing > and afterwards when it starts playing again. This can be fixed by > disabling power save on the audio driver [...]. I've only written this patch because the issue seemed similar to that on my Lenovo Yoga 9 14IAP7. sound/pci/hda/patch_realtek.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 38930cf5aace..e6f229e418c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6896,6 +6896,32 @@ static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, } } +static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170151 }, + { 0x17, 0x90170150 }, + { } + }; + static const hda_nid_t conn[] = { 0x02, 0x03 }; + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x03, + 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7137,6 +7163,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9066,6 +9093,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_dell_inspiron_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9165,6 +9198,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.3

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[PATCH] ALSA: hda/realtek: Enable 4-speaker output Dell Precision 5560 laptop
by acheronfail 14 Sep '22

14 Sep '22

This is my first patch, so let me know if I need to do anything differently. :) The Dell Precision 5570 uses the same 4-speakers-on-ALC289 just like the previous Precision 5560. I replicated that patch onto this one, and can confirm that the audio is much better (the woofers are now working); I've tested it on my Dell Precision 5570. --- 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 799f6bf266dd..b0ad4856b877 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9130,6 +9130,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.3

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RE: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp
by Ki-Seok Jo 14 Sep '22

14 Sep '22

Dear Mark Brown, Thanks for your detailed feedback. I'm so late in replying. COVID-19 is going around again in Korea. I also suffered from COVID-19 this time. Please take care of yourself from COVID-19. Anyway, I proceeded with the update again by referring to your review. And I added my opinion to the following. Please check the below. If there is anything wrong, please give me a feedback again. Thank you for your patience and cooperation. Best regards, Ki Seok Jo Iron Device Corporation Tel.: +82-2-541-2896 Mobile: +82-10-3320-0376 Email: kiseok.jo(a)irondevice.com The information in this email and any attachment is confidential and may be legally protected. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure or actions taken as a result of the information in this email is prohibited and may be unlawful. If you are not the intended recipient, please notify us immediately and then delete this e-mail and any attachment. -----Original Message----- From: Mark Brown <broonie(a)kernel.org> Sent: Wednesday, August 17, 2022 9:09 PM To: Ki-Seok Jo <kiseok.jo(a)irondevice.com> Cc: Gyu-Hwa Park <gyuhwa.park(a)irondevice.com>; alsa-devel(a)alsa-project.org; Suk-Min Kang <sukmin.kang(a)irondevice.com> Subject: Re: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp On Wed, Aug 17, 2022 at 12:29:37PM +0900, Kiseok Jo wrote: I got part way through reviewing this driver. The top level feedback here is that the driver appears to be exposing everything in the device to userspace as enumeration controls rather than using standard features like DAPM, or appropriate other control types where there's something that's a better fit than an enum. If there's a strong reason for the driver to do something so unusual then theere should be something in here which explains what's going on. # To solve the any issue by changing the value whenever there is a problem during use, we registered to control all functions. # Since the verification was completed by mass production with the corresponding setting, we removed the controls in the userspace except for a few things as you mentioned. > +// SPDX-License-Identifier: GPL-2.0-or-later > +/* sma1303.c -- sma1303 ALSA SoC Audio driver > + * > + * Copyright 2019 Iron Device Corporation 2019? # I missed this point. Thank you! # Changed 2019 to 2022. Please make the entire block a C++ one so things look more intentional. # I have modified the code a bit. I'm sorry, but I couldn't figure out exactly what you were talking about. # If there is anything that needs to be corrected, please reply again. > +static int power_up_down_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + if (sel && !(sma1303->force_amp_power_down)) > + sma1303_startup(component); > + else > + sma1303_shutdown(component); > + > + return 0; > +} This and all the other controls need to return 1 on change, please make sure a card with your driver your driver passes mixer-test cleanly - it will spot this and other problems. # Added the return value for checking error. # It's ok if the value is '0', and if a problem occurs, it returns to a value of less than 0. # I have one question. All return values have to match the Linux standard error code? However I wouldn't expect the power controls to exist at all, generally power is managed via DAPM rather than userspace. # Removed the mixer control to control power from userspace. # Automatically power control as DAC Event(DAPM), bias, set hw parameter and so on.. > +static int force_sdo_bypass_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + sma1303->sdo_bypass_flag = (bool)sel; > + > + return 0; > +} What is this bypassing? # In chip, this is a function to set not to output on I2S. # The output will always be on. # So, We removed this. > +static const char * const sma1303_input_format_text[] = { > + "I2S", "LJ", "Reserved", "Reserved", > + "RJ_16", "RJ_18", "RJ_20", "RJ_24"}; > + > +static const struct soc_enum sma1303_input_format_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_format_text), > + sma1303_input_format_text); This should be controlled by the machine driver through set_dai_fmt() and hw_params(). # It's already included in that. # This mixer control is an additional control for debugging. # So, We removed this. > +static const char * const sma1303_pcm_n_slot_text[] = { > + "Slot_1", "Slot_2", "Slot_3", "Slot_4", "Slot_5", "Slot_6", > + "Slot_7", "Slot_8", "Slot_9", "Slot_10", "Slot_11", "Slot_12", > + "Slot_13", "Slot_14", "Slot_15", "Slot_16"}; > + > +static const struct soc_enum sma1303_pcm_n_slot_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_pcm_n_slot_text), > + sma1303_pcm_n_slot_text); set_tdm_slot() # It's already included in that. # This mixer control is also an additional control for debugging. # So, We removed this. > +static const char * const sma1303_port_config_text[] = { > + "IN_Port", "Reserved", "OUT_Port", "Reserved"}; This should be DT properties, possibly using the pinctrl schema depending on what's actually going on. # I'm sorry. I think I named it wrong. # This control is a feature that sets whether output is enabled or not. Output means SDO, that is, data exported from the chip. # So I modified this. > +static const char * const sma1303_port_out_sel_text[] = { > + "Disable", "Format_C", "Mixer_out", "After_DSP", > + "Postscaler", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_port_out_sel_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_port_out_sel_text), > + sma1303_port_out_sel_text); This looks like it should be a DAPM routing control. # That's right. I added the control in route. > +static const char * const sma1303_spk_off_slope_text[] = { > + "00", "01", "10", "11"}; > + > +static const struct soc_enum sma1303_spk_off_slope_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spk_off_slope_text), > + sma1303_spk_off_slope_text); Why make this an enum and not just a numerical control with values 0-3? # Yes. I missed this value. # This function has been removed as users do not need to control it > +static const char * const sma1303_spkmode_text[] = { > + "Off", "Mono", "Reserved", "Reserved", > + "Stereo", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_spkmode_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spkmode_text), > + sma1303_spkmode_text); Use a value enum to eliminate the reserved values from user selection, however if this is controlling if the device has mono or stereo speakers it should be a DT property since that's unlikely to change at runtime. # This function also includes PWM on/off. # We have changed the enum. > +static const char * const sma1303_input_gain_text[] = { > + "Gain_0dB", "Gain_M6dB", "Gain_M12dB", "Gain_MInf"}; > + > +static const struct soc_enum sma1303_input_gain_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_gain_text), > + sma1303_input_gain_text); This should just be a normal volume control. # We have other control for normal volume. # So, this control has been removed. I've stopped reviewing at this point. # Thanks for the detailed feedback.

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[PATCH v1] ASoC: rockchip: i2s: use regmap_read_poll_timeout to poll I2S_CLR
by Judy Hsiao 14 Sep '22

14 Sep '22

Use regmap_read_poll_timeout to poll I2S_CLR. It also fixes the 'rockchip-i2s ff070000.i2s; fail to clear' and return -EBUSY when the read of I2S_CLR exceeds the retry limit. Fixes: 0ff9f8b9f592 ("ASoC: rockchip: i2s: Fix error code when fail to read I2S_CLR") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 39 +++++++++++++------------------ 1 file changed, 16 insertions(+), 23 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index f5f3540a9e18..02905eec615e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -163,18 +163,14 @@ static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_CLR_TXC | I2S_CLR_RXC); if (ret < 0) goto end; - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - ret = -EBUSY; - break; - } - } + ret = regmap_read_poll_timeout(i2s->regmap, + I2S_CLR, + val, + val != 0, + 20, + 200); + if (ret < 0) + dev_warn(i2s->dev, "fail to clear\n"); } } end: @@ -226,17 +222,14 @@ static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_CLR_TXC | I2S_CLR_RXC); if (ret < 0) goto end; - regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - ret = -EBUSY; - break; - } - } + ret = regmap_read_poll_timeout(i2s->regmap, + I2S_CLR, + val, + val != 0, + 20, + 200); + if (ret < 0) + dev_warn(i2s->dev, "fail to clear\n"); } } end: -- 2.37.2.789.g6183377224-goog

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