March 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: sgio2audio: Remove usage of dropped hw_params/hw_free functions
by Thomas Bogendoerfer 06 Mar '20

06 Mar '20

Commit ee88f4ebe575 ("ALSA: mips: Use managed buffer allocation") removed superfluous hw_params/hw_free callbacks, but forgot to remove them where they were used. Fixes: ee88f4ebe575 ("ALSA: mips: Use managed buffer allocation") Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- sound/mips/sgio2audio.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9f60a5037f8b..5bf1ea150f26 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -649,8 +649,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .open = snd_sgio2audio_playback1_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, @@ -659,8 +657,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .open = snd_sgio2audio_playback2_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, @@ -669,8 +665,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .open = snd_sgio2audio_capture_open, .close = snd_sgio2audio_pcm_close, - .hw_params = snd_sgio2audio_pcm_hw_params, - .hw_free = snd_sgio2audio_pcm_hw_free, .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, -- 2.25.0

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strange loopback behaviour seen in alsamixer
by Dr. Jochen Blödorn 06 Mar '20

06 Mar '20

hi, while debugging my sound setup in Xubuntu 18.4, I came up on some strange behaviour. Reason for debugging was, I couldn't play rear line input to line output. To check, if I missed some basic settings I started alsamixer. As I always do, I toggled each setting, to make sure, that the displayed status reflects it's actual state. This routine is one relict from my old windows times :-) When I applied this method to "loopback", which initially was displayed as "enabled" ( switch from enabled to disabled and back to enabled), I instantly got sound output. This can, and must be, reproduced after every restart of the desktop. Sorry, if I posted this mail wrong way, wrong place, but I thought, the observation might be of interest. Jochen

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Automatically switching snd_hda_intel.power_save value when switching from battery to ac ?
by Hans de Goede 06 Mar '20

06 Mar '20

Hi, Because of a bug-report about power-saving related plops/clocks on a Lenovo T470s, I've asked inside Red Hat if people with a T470s and running a recent kernel were also experiencing this. Most people are happy with the audio, but I did get a few bug reports about plops on the headphones-jack. One of the suggestions which I got from 2 different users is to disable power-saving for the HDA driver when on AC, esp. since most headphones-jack use (esp. with an external amplifier which amplifies the problem) is done while the laptop is sitting on a desk and thus typically is connected to a charger. I'm personally not necessarily a fan of changing settings based on being connected to ac or not, but I guess that in this case it might not be such a bad idea ? Regards, Hans

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[PATCH] ALSA: usb-audio: Fix missing braces in some struct inits
by Takashi Iwai 06 Mar '20

06 Mar '20

The struct s1810c_state_packet contains the array in the first field hence zero-initialization requires a more couple of braces. Fix the compile warning pointing it out: sound/usb/mixer_s1810c.c: In function 'snd_sc1810c_get_status_field': sound/usb/mixer_s1810c.c:178:9: warning: missing braces around initializer [-Wmissing-braces] Reported-by: kbuild test robot <lkp(a)intel.com> Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c") Link: https://lore.kernel.org/r/202002210251.WgMfvKJP%lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_s1810c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 816879a07f82..6483e47bafd0 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -175,8 +175,8 @@ static int snd_sc1810c_get_status_field(struct usb_device *dev, u32 *field, int field_idx, uint16_t *seqnum) { - struct s1810c_state_packet pkt_out = { 0 }; - struct s1810c_state_packet pkt_in = { 0 }; + struct s1810c_state_packet pkt_out = { { 0 } }; + struct s1810c_state_packet pkt_in = { { 0 } }; int ret = 0; pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; -- 2.16.4

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[PATCH] ALSA: usb-audio: Add support for MOTU MicroBook IIc
by Alexander Tsoy 06 Mar '20

06 Mar '20

MicroBook IIc operates in UAC2 mode by default. This patch addresses several issues with it: - MicroBook II and IIc shares the same USB ID. We can distinguish them by interface class. - MaxPacketsOnly attribute is erroneously set in endpoint descriptors. As a result this card produces noise with all sample rates other than 96 KHz. This also causes issues like IOMMU page faults and other problems with host controller. - Sample rate changes takes more than 2 seconds for this device. Clock validity request returns false during that period, so the clock validity quirk is required. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 59 ++++++++++++++++++++++++++++++++-------- sound/usb/pcm.c | 7 ++++- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 18 +++++++++++- 4 files changed, 72 insertions(+), 14 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a48313dfa967..b118cf97607f 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } -/* - * Assume the clock is valid if clock source supports only one single sample - * rate, the terminal is connected directly to it (there is no clock selector) - * and clock type is internal. This is to deal with some Denon DJ controllers - * that always reports that clock is invalid. - */ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, int source_id) { + bool ret = false; + int count; + unsigned char data; + struct usb_device *dev = chip->dev; + if (fmt->protocol == UAC_VERSION_2) { struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); @@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, if (!cs_desc) return false; - return (fmt->nr_rates == 1 && - (fmt->clock & 0xff) == cs_desc->bClockID && - (cs_desc->bmAttributes & 0x3) != - UAC_CLOCK_SOURCE_TYPE_EXT); + /* + * Assume the clock is valid if clock source supports only one + * single sample rate, the terminal is connected directly to it + * (there is no clock selector) and clock type is internal. + * This is to deal with some Denon DJ controllers that always + * reports that clock is invalid. + */ + if (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT) + return true; + } + + /* + * MOTU MicroBook IIc + * Sample rate changes takes more than 2 seconds for this device. Clock + * validity request returns false during that period. + */ + if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { + count = 0; + + while ((!ret) && (count < 50)) { + int err; + + msleep(100); + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data)); + if (err < 0) { + dev_warn(&dev->dev, + "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return false; + } + + ret = !!data; + count++; + } } - return false; + return ret; } static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bd258f1ec2dd..a4e4064f9aee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; - case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */ + /* MicroBook IIc */ + if (altsd->bInterfaceClass == USB_CLASS_AUDIO) + return 0; + + /* MicroBook II */ ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..1c8719292eee 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, /* MOTU Microbook II */ { - USB_DEVICE(0x07fd, 0x0004), + USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MOTU", .product_name = "MicroBookII", diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7f558f4b4520..a7bd32bd301c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1316,7 +1316,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ - return snd_usb_motu_microbookii_boot_quirk(dev); + /* + * For some reason interface 3 with vendor-spec class is + * detected on MicroBook IIc. + */ + if (get_iface_desc(intf->altsetting)->bInterfaceClass == + USB_CLASS_VENDOR_SPEC && + get_iface_desc(intf->altsetting)->bInterfaceNumber < 3) + return snd_usb_motu_microbookii_boot_quirk(dev); + break; } return 0; @@ -1754,5 +1762,13 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, else fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */ + /* + * MaxPacketsOnly attribute is erroneously set in endpoint + * descriptors. As a result this card produces noise with + * all sample rates other than 96 KHz. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; + break; } } -- 2.24.1

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Re: [PATCH v4 10/10] sound: bt87x: use pci_status_get_and_clear_errors
by Takashi Iwai 06 Mar '20

06 Mar '20

On Sat, 29 Feb 2020 23:29:07 +0100, Heiner Kallweit wrote: > > Use new helper pci_status_get_and_clear_errors() to simplify the code. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi > --- > sound/pci/bt87x.c | 7 +------ > 1 file changed, 1 insertion(+), 6 deletions(-) > > diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c > index 8c48864c8..656750466 100644 > --- a/sound/pci/bt87x.c > +++ b/sound/pci/bt87x.c > @@ -271,13 +271,8 @@ static void snd_bt87x_free_risc(struct snd_bt87x *chip) > > static void snd_bt87x_pci_error(struct snd_bt87x *chip, unsigned int status) > { > - u16 pci_status; > + int pci_status = pci_status_get_and_clear_errors(chip->pci); > > - pci_read_config_word(chip->pci, PCI_STATUS, &pci_status); > - pci_status &= PCI_STATUS_PARITY | PCI_STATUS_SIG_TARGET_ABORT | > - PCI_STATUS_REC_TARGET_ABORT | PCI_STATUS_REC_MASTER_ABORT | > - PCI_STATUS_SIG_SYSTEM_ERROR | PCI_STATUS_DETECTED_PARITY; > - pci_write_config_word(chip->pci, PCI_STATUS, pci_status); > if (pci_status != PCI_STATUS_DETECTED_PARITY) > dev_err(chip->card->dev, > "Aieee - PCI error! status %#08x, PCI status %#04x\n", > -- > 2.25.1 > >

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[PATCH v2 0/1] Fix silent output on Gigabyte X570 Aorus Master
by Christian Lachner 06 Mar '20

06 Mar '20

The Gigabyte X570 Aorus Master motherboard with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. This patch sets up a quirk entry for that. It was tested by myself on my own hardware for some time and it works just fine. We should probably rename ALC1220_FIXUP_CLEVO_P950 at some point as the amount of non-Clevo hardware needing this workaround is growing. Christian Lachner (1): ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) base-commit: 98d54f81e36ba3bf92172791eba5ca5bd813989b -- 2.25.1

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[PATCH] sound: korg1212: fix if-statement empty body warnings
by Randy Dunlap 06 Mar '20

06 Mar '20

From: Randy Dunlap <rdunlap(a)infradead.org> Fix gcc warnings when -Wextra is used by using an empty do-while block instead of <nothing>. Fixes these build warnings: ../sound/pci/korg1212/korg1212.c:674:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:708:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:730:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:853:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1013:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1035:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1052:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1066:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1087:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1094:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:1208:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] ../sound/pci/korg1212/korg1212.c:2360:102: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/korg1212/korg1212.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200225.orig/sound/pci/korg1212/korg1212.c +++ linux-next-20200225/sound/pci/korg1212/korg1212.c @@ -30,7 +30,7 @@ #if K1212_DEBUG_LEVEL > 0 #define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) #else -#define K1212_DEBUG_PRINTK(fmt,...) +#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0) #endif #if K1212_DEBUG_LEVEL > 1 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args)

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Get cross compilation to work
by GitHub pull_request - opened 06 Mar '20

06 Mar '20

alsa-project/alsa-firmware pull request #1 was opened from matthewbauer: Rebased version of original patch by Ding Xiang Fei found in https://github.com/NixOS/nixpkgs/pull/63866. This version applies in alsa-firmware-1.2.1. Co-authored-by: Ding Xiang Fei <dingxiangfei2009(a)users.noreply.github.com> Request URL : https://github.com/alsa-project/alsa-firmware/pull/1 Patch URL : https://github.com/alsa-project/alsa-firmware/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-firmware

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[PATCH] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_ssm4567 machine
by Radoslaw Biernacki 05 Mar '20

05 Mar '20

From: Ben Zhang <benzh(a)chromium.org> This single fix address two issues on machine with nau88125 + ssm4567: 1) Audio distortion, due to lack of required clock rate on MCLK line 2) Loud audible "pops" on headphones if there is no sysclk during nau8825 playback power up sequence Explanation for: 1) Due to Skylake HW limitation, MCLK pin can only output 24MHz clk rate (it can be only connected to XTAL parent clk). The BCLK pin can be driven by dividers and therefore FW is able to set it to rate required by chosen audio format. According to nau8825 datasheet, 256*FS sysclk gives the best audio quality and the only way to achieve this (taking into account the above limitations) its to regenerate the MCLK from BCLK on nau8825 side by FFL. Without required clk rate, audio is distorted by added harmonics. 2) Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by letting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. Once device resumes, the system will only enable power sequence for playback without doing hardware parameter, audio format, and PLL configure. In the mean time, the jack detecion sequence has changed PLL parameters and switched to internal clock. Thus, the playback signal distorted without correct PLL parameters. That is why we need to configure the PLL again in SNDRV_PCM_TRIGGER_RESUME case. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 71 ++++++++++++++----- 1 file changed, 52 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509..c5354da49670 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,7 +58,7 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,40 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + msleep(20); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { @@ -671,10 +682,32 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->playback_active, codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->playback_active && codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, -- 2.17.1

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