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[PATCH AUTOSEL 5.14 08/40] ALSA: usb-audio: Unify mixer resume and reset_resume procedure
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 7b9cf9036609428e845dc300aec13822ba2c4ab3 ] USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu <enshuo(a)chromium.org> Cc: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsM… Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 6abfc9d079e7..fa75b7e72ad1 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1020,7 +1020,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1046,7 +1046,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1066,20 +1066,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1101,7 +1091,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9b713b4a5ec4..fa7cf982d39e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3655,33 +3655,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3702,7 +3685,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ea41e7a1f7bf..16567912b998 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -122,7 +121,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7d00..4a4d3361ac04 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); -- 2.33.0

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[PATCH AUTOSEL 5.14 05/40] ALSA: oxfw: fix transmission method for Loud models based on OXFW971
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 64794d6db49730d22f440aef0cf4da98a56a4ea3 ] Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210913021042.10085-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } -- 2.33.0

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[PATCH AUTOSEL 5.14 03/40] ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic
by Sasha Levin 05 Oct '21

05 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 58eafe1ff52ee1ce255759fc15729519af180cbb ] The SoundWire BEs make use of 'stream' functions for .prepare and .trigger. These functions will in turn force a Bank Switch, which implies a wait operation. Mark SoundWire BEs as nonatomic for consistency, but keep all other types of BEs as is. The initialization of .nonatomic is done outside of the create_sdw_dailink helper to avoid adding more parameters to deal with a single exception to the rule that BEs are atomic. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20210907184436.33152-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1a867c73a48e..cb3afc4519cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -860,6 +860,11 @@ static int create_sdw_dailink(struct device *dev, int *be_index, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(link, dai_links + (*be_index)++, playback, group_id); -- 2.33.0

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Re: [PATCH v2 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:41:31AM +0800, cy_huang wrote: > Very appoligized. I forget to migrate this line from my test 5.10 kernel. > Without this line will cause sound card codec link fail. > Please use the patch v2 to review the bindings and source code. Sorry, saw v1 before I saw this but I guess the issues still apply - a quick check seems like it.

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Re: [PATCH 2/2] ASoC: rt9120: Add rt9210 audio amplifier support
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:55AM +0800, cy_huang wrote: > +static const char * const sdo_select_text[] = { > + "NONE", "INTF", "FINAL", "RMS Detect" > +}; Why not None and Final? > + if (event == SND_SOC_DAPM_PRE_PMU) > + snd_soc_component_write(comp, RT9120_REG_ERRRPT, 0); > + else if (event == SND_SOC_DAPM_POST_PMU) > + msleep(RT9120_AMPON_WAITMS); > + else if (event == SND_SOC_DAPM_POST_PMD) > + msleep(RT9120_AMPOFF_WAITMS); This should be a switch statement, it'd be clearer. > + /* Default config volume to 0dB */ > + snd_soc_component_write(comp, RT9120_REG_MSVOL, 0x180); > + /* Mute by default */ > + snd_soc_component_update_bits(comp, RT9120_REG_VOLRAMP, > + RT9120_MUTE_MASK, RT9120_MUTE_MASK); As ever you should leave the defaults at whatever the hardware defaults to, the defaults for one machine may not be suitable for another so we shouldn't be trying to do that in software.

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Re: [PATCH 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:54AM +0800, cy_huang wrote: > + richtek,use-dvdd-1p8v: > + description: Indicate DVDD 1P8V is used, default for 3P3V or 5V design > + type: boolean I would expect this to be done through the regulator bindings, they would allow the driver to query the supply voltage.

1 0

[PATCH] ASoC: rt5682: move clk related code to rt5682_i2c_probe
by Jack Yu 05 Oct '21

05 Oct '21

The DAI clock is only used in I2S mode, to make it clear and to fix clock resource release issue, we move CCF clock related code to rt5682_i2c_probe to fix clock register/unregister issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5682-i2c.c | 22 +++++++++++ sound/soc/codecs/rt5682.c | 70 +++++++++++++---------------------- sound/soc/codecs/rt5682.h | 3 ++ 3 files changed, 51 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index b9d5d7a0975b..a30e42932cf7 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -139,6 +139,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5682); + rt5682->i2c_dev = &i2c->dev; + rt5682->pdata = i2s_default_platform_data; if (pdata) @@ -276,6 +278,26 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(rt5682); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = 48000; +#endif + return devm_snd_soc_register_component(&i2c->dev, &rt5682_soc_component_dev, rt5682_dai, ARRAY_SIZE(rt5682_dai)); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 12113c2dcae2..914fe7debc05 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2510,7 +2510,7 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, static bool rt5682_clk_check(struct rt5682_priv *rt5682) { if (!rt5682->master[RT5682_AIF1]) { - dev_dbg(rt5682->component->dev, "sysclk/dai not set correctly\n"); + dev_dbg(rt5682->i2c_dev, "sysclk/dai not set correctly\n"); return false; } return true; @@ -2521,13 +2521,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); + struct snd_soc_component *component; + struct snd_soc_dapm_context *dapm; if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); @@ -2557,13 +2559,15 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); + struct snd_soc_component *component; + struct snd_soc_dapm_context *dapm; if (!rt5682_clk_check(rt5682)) return; + component = rt5682->component; + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); @@ -2587,7 +2591,6 @@ static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) @@ -2597,7 +2600,7 @@ static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, */ if (rt5682->lrck[RT5682_AIF1] != CLK_48 && rt5682->lrck[RT5682_AIF1] != CLK_44) { - dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + dev_warn(rt5682->i2c_dev, "%s: clk %s only support %d or %d Hz output\n", __func__, clk_name, CLK_44, CLK_48); return 0; } @@ -2611,7 +2614,6 @@ static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) @@ -2621,7 +2623,7 @@ static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, * It will force to 48kHz if not both. */ if (rate != CLK_48 && rate != CLK_44) { - dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + dev_warn(rt5682->i2c_dev, "%s: clk %s only support %d or %d Hz output\n", __func__, clk_name, CLK_44, CLK_48); rate = CLK_48; } @@ -2635,7 +2637,7 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; + struct snd_soc_component *component; struct clk_hw *parent_hw; const char * const clk_name = clk_hw_get_name(hw); int pre_div; @@ -2644,6 +2646,8 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + /* * Whether the wclk's parent clk (mclk) exists or not, please ensure * it is fixed or set to 48MHz before setting wclk rate. It's a @@ -2653,12 +2657,12 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, */ parent_hw = clk_hw_get_parent(hw); if (!parent_hw) - dev_warn(component->dev, + dev_warn(rt5682->i2c_dev, "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", CLK_PLL2_FIN); if (parent_rate != CLK_PLL2_FIN) - dev_warn(component->dev, "clk %s only support %d Hz input\n", + dev_warn(rt5682->i2c_dev, "clk %s only support %d Hz input\n", clk_name, CLK_PLL2_FIN); /* @@ -2690,10 +2694,9 @@ static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_BCLK_IDX]); - struct snd_soc_component *component = rt5682->component; unsigned int bclks_per_wclk; - bclks_per_wclk = snd_soc_component_read(component, RT5682_TDM_TCON_CTRL); + regmap_read(rt5682->regmap, RT5682_TDM_TCON_CTRL, &bclks_per_wclk); switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { case RT5682_TDM_BCLK_MS1_256: @@ -2754,20 +2757,22 @@ static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_BCLK_IDX]); - struct snd_soc_component *component = rt5682->component; + struct snd_soc_component *component; struct snd_soc_dai *dai; unsigned long factor; if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + factor = rt5682_bclk_get_factor(rate, parent_rate); for_each_component_dais(component, dai) if (dai->id == RT5682_AIF1) break; if (!dai) { - dev_err(component->dev, "dai %d not found in component\n", + dev_err(rt5682->i2c_dev, "dai %d not found in component\n", RT5682_AIF1); return -ENODEV; } @@ -2790,10 +2795,9 @@ static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { }, }; -static int rt5682_register_dai_clks(struct snd_soc_component *component) +int rt5682_register_dai_clks(struct rt5682_priv *rt5682) { - struct device *dev = component->dev; - struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct device *dev = rt5682->i2c_dev; struct rt5682_platform_data *pdata = &rt5682->pdata; struct clk_hw *dai_clk_hw; int i, ret; @@ -2851,6 +2855,7 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) return 0; } +EXPORT_SYMBOL_GPL(rt5682_register_dai_clks); #endif /* CONFIG_COMMON_CLK */ static int rt5682_probe(struct snd_soc_component *component) @@ -2860,9 +2865,6 @@ static int rt5682_probe(struct snd_soc_component *component) unsigned long time; struct snd_soc_dapm_context *dapm = &component->dapm; -#ifdef CONFIG_COMMON_CLK - int ret; -#endif rt5682->component = component; if (rt5682->is_sdw) { @@ -2874,26 +2876,6 @@ static int rt5682_probe(struct snd_soc_component *component) dev_err(&slave->dev, "Initialization not complete, timed out\n"); return -ETIMEDOUT; } - } else { -#ifdef CONFIG_COMMON_CLK - /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(rt5682->mclk)) { - if (PTR_ERR(rt5682->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682->mclk); - return ret; - } - rt5682->mclk = NULL; - } - - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; - - /* Initial setup for CCF */ - rt5682->lrck[RT5682_AIF1] = CLK_48; -#endif } snd_soc_dapm_disable_pin(dapm, "MICBIAS"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b59221048ebf..262512babc50 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1408,6 +1408,7 @@ enum { struct rt5682_priv { struct snd_soc_component *component; + struct device *i2c_dev; struct rt5682_platform_data pdata; struct regmap *regmap; struct regmap *sdw_regmap; @@ -1462,6 +1463,8 @@ void rt5682_calibrate(struct rt5682_priv *rt5682); void rt5682_reset(struct rt5682_priv *rt5682); int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev); +int rt5682_register_dai_clks(struct rt5682_priv *rt5682); + #define RT5682_REG_NUM 318 extern const struct reg_default rt5682_reg[RT5682_REG_NUM]; -- 2.33.0

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User unsubscribed from mailing list
by Manish DUBEY 05 Oct '21

05 Oct '21

Hello Team, Hope you are doing well. We have some users in ST environment being subscribed to your Mailing list. After few days they use to get an email stating that "Your membership in the mailing list has been disabled due to excessive bounces".. We have verified at our end and didn't find any bounce back email getting generated. To investigate the case, could you please share couple of sample bounce back (NDR) email received by you from ST environment. The NDR may help us to investigate the cause of bounce back email getting generated. If possible, we can schedule a call as well to discuss the case. Feel free to share your availability to discuss the case and we can share a Teams invite accordingly. Awaiting your response. [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 ST Restricted

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User unsubscribed from mailing list
by Manish DUBEY 05 Oct '21

05 Oct '21

Hello Team, Hope you are doing well. We have some users in ST environment being subscribed to your Mailing list. After few days they use to get an email stating that "Your membership in the mailing list has been disabled due to excessive bounces".. We have verified at our end and didn't find any bounce back email getting generated. To investigate the case, could you please share couple of sample bounce back (NDR) email received by you from ST environment. The NDR may help us to investigate the cause of bounce back email getting generated. If possible, we can schedule a call as well to discuss the case. Feel free to share your availability to discuss the case and we can share a Teams invite accordingly. Awaiting your response. [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 ST Restricted

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[PATCH] ASoC: qcom: apq8096: Constify static snd_soc_ops
by Rikard Falkeborn 05 Oct '21

05 Oct '21

The struct iapq8096_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/qcom/apq8096.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1a69baefc5ce..c7b7d0864d1a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -60,7 +60,7 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops apq8096_ops = { +static const struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, }; -- 2.33.0

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