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)

2 1

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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[PATCH] ALSA: hda/realtek - Enable the headset of ASUS B9450FA with ALC294
by Jian-Hong Pan 05 Mar '20

05 Mar '20

A headset on the laptop like ASUS B9450FA does not work, until quirk ALC294_FIXUP_ASUS_HPE is applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Singed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 477589e7ec1d..a47f6404aea9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5920,7 +5920,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - + ALC294_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7040,7 +7040,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - + [ALC294_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7204,6 +7214,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.25.1

2 1

Re: [PATCH 1/1] ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master
by Takashi Iwai 05 Mar '20

05 Mar '20

On Sun, 23 Feb 2020 10:24:16 +0100, Christian Lachner wrote: > > The Gigabyte X570 Aorus Master motherboard with ALC1220 codec > requires a similar workaround for Clevo laptops to enforce the > DAC/mixer connection path. Set up a quirk entry for that. > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 > Signed-off-by: Christian Lachner <gladiac(a)gmail.com> Applied now. Thanks. Takashi

1 0

Enable Headset Button for ThinkPad X1
by Kailang 05 Mar '20

05 Mar '20

Hi Takashi, Lenovo ThinkPad X1 want to support Headset Button. Attach patch will enable it. BR, Kailang

2 1

RE: Enable Headset Mic for Dell desktop
by Kailang 05 Mar '20

05 Mar '20

Sorry! Forgot attach file. > -----Original Message----- > From: Kailang > Sent: Thursday, February 20, 2020 4:30 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Enable Headset Mic for Dell desktop > > Hi Takashi, > > Attach patch was supported headset Mic. > > BR, > Kailang

2 1

snd_hda_intel.power_save=1 causes pops/clicks on Lenovo T470s headphone jack?
by Hans de Goede 05 Mar '20

05 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. Is this a know issue? and/or is there anything we can do about this? Regards, Hans

1 1

[PATCH for-next] ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification
by Dan Murphy 05 Mar '20

05 Mar '20

Fix the range verification check for the mic_bias and vref device tree entries. Fixes 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662c..38897568ee96 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -748,9 +748,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) bias_source = ADCX140_MIC_BIAS_VAL_VREF; - if (bias_source != ADCX140_MIC_BIAS_VAL_VREF && - bias_source != ADCX140_MIC_BIAS_VAL_VREF_1096 && - bias_source != ADCX140_MIC_BIAS_VAL_AVDD) { + if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || + bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } @@ -760,9 +759,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) vref_source = ADCX140_MIC_BIAS_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { + if (vref_source < ADCX140_MIC_BIAS_VREF_275V || + vref_source > ADCX140_MIC_BIAS_VREF_1375V) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } -- 2.25.1

2 1

[PATCH] ASoC: sprd: Allow the MCDT driver to build into modules
by Baolin Wang 05 Mar '20

05 Mar '20

Change the config to 'tristate' for MCDT driver to allow it to build into modules, as well as changing to use IS_ENABLED() to validate if need supply dummy functions when building the MCDT driver as a module. Signed-off-by: Baolin Wang <baolin.wang7(a)gmail.com> --- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 5474fd3..5e0ac82 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -8,7 +8,7 @@ config SND_SOC_SPRD the Spreadtrum SoCs' Audio interfaces. config SND_SOC_SPRD_MCDT - bool "Spreadtrum multi-channel data transfer support" + tristate "Spreadtrum multi-channel data transfer support" depends on SND_SOC_SPRD help Say y here to enable multi-channel data transfer support. It diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h index 9cc7e207..679e3af 100644 --- a/sound/soc/sprd/sprd-mcdt.h +++ b/sound/soc/sprd/sprd-mcdt.h @@ -48,7 +48,7 @@ struct sprd_mcdt_chan { struct list_head list; }; -#ifdef CONFIG_SND_SOC_SPRD_MCDT +#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT) struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type); void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); -- 1.9.1

2 1

Applied "ASoC: soc-pcm: use defined stream" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: use defined stream has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 25c2f5156dd57f03aee2de079248c23a56222c92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:38 +0900 Subject: [PATCH] ASoC: soc-pcm: use defined stream Many functions defines "stream = substream->stream", but some of them is using "substream->stream" instead of "stream". It is pointless. This patch uses defined stream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87mu947q1t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 90857138c823..8c27eb4d5e4c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -644,8 +644,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(codec_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); @@ -2149,7 +2148,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - ret = dpcm_be_dai_startup(fe, fe_substream->stream); + ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret); goto be_err; @@ -2180,7 +2179,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) return 0; unwind: - dpcm_be_dai_startup_unwind(fe, fe_substream->stream); + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; @@ -2234,7 +2233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ - dpcm_be_dai_shutdown(fe, substream->stream); + dpcm_be_dai_shutdown(fe, stream); dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); @@ -2412,9 +2411,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - memcpy(&fe->dpcm[substream->stream].hw_params, params, + memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); - ret = dpcm_be_dai_hw_params(fe, substream->stream); + ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; @@ -2736,7 +2735,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - ret = dpcm_be_dai_prepare(fe, substream->stream); + ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; -- 2.20.1

1 0

Applied "ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6e02feb0d2663c1b7caa5e271c2a60e219f0ca07 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:48 +0900 Subject: [PATCH] ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() dpcm_add_paths() checks returned be from dpcm_get_be() static int dpcm_add_paths(...) { ... for_each_dapm_widgets(list, i, widget) { ... be = dpcm_get_be(...); ... /* make sure BE is a real BE */ => if (!be->dai_link->no_pcm) continue; ... } ... } But, dpcm_get_be() itself is checking it already. dpcm_get_be(...) { ... for_each_card_rtds(card, be) { => if (!be->dai_link->no_pcm) continue; ... if (...) => return be; } return NULL } This patch removes duplicate check Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87lfoo7q1j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8c27eb4d5e4c..e3a2c4f7757b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1690,10 +1690,6 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* make sure BE is a real BE */ - if (!be->dai_link->no_pcm) - continue; - /* don't connect if FE is not running */ if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; -- 2.20.1

1 0

sof-hda-dsp: Support systems without integrated graphics audio
by GitHub pull_request - opened 05 Mar '20

05 Mar '20

alsa-project/alsa-ucm-conf pull request #15 was opened from kv2019i: On systems where integrated graphics audio is not present or is disabled, the HDMI PCM nodes are disabled. Add rules to detect these systems by checking presence of HDMI jack controls with UCM2 rules. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/8] ASoC: soc-pcm cleanup step5
by Kuninori Morimoto 05 Mar '20

05 Mar '20

Hi Mark These are v2 of soc-pcm cleanup step5. These are based on latest for-5.7 branch. Mainly v2 fixed the points reviewed in v1. [7/8] patch might exchange the behavior Kuninori Morimoto (8): 1) ASoC: soc-pcm: use defined stream 2) ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() 3) ASoC: soc-pcm: move dpcm_fe_dai_close() 4) ASoC: soc-pcm: add dpcm_fe_dai_cleanup() 5) ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() 6) ASoC: soc-pcm: check DAI's activity more simply 7) ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. 8) ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() sound/soc/soc-pcm.c | 114 +++++++++++++++++++------------------------- 1 file changed, 50 insertions(+), 64 deletions(-) -- 2.17.1

2 12

[PATCH] ASoC: wm8741: Fix typo in Kconfig prompt
by Axel Lin 05 Mar '20

05 Mar '20

Fix trivial copy-n-paste mistake. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6aee70ed43df..fd09656d2f6b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1486,7 +1486,7 @@ config SND_SOC_WM8737 depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate "Wolfson Microelectronics WM8737 DAC" + tristate "Wolfson Microelectronics WM8741 DAC" depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 -- 2.20.1

2 2

[PATCH] ASoc: amd: acp3x: Add missing include <linux/io.h>
by YueHaibing 05 Mar '20

05 Mar '20

gcc 7.4.0 build fails: In file included from sound/soc/amd/acp3x-rt5682-max9836.c:20:0: sound/soc/amd/raven/acp3x.h: In function rv_readl: sound/soc/amd/raven/acp3x.h:113:9: error: implicit declaration of function readl; did you mean rv_readl? [-Werror=implicit-function-declaration] return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~ rv_readl sound/soc/amd/raven/acp3x.h: In function rv_writel: sound/soc/amd/raven/acp3x.h:118:2: error: implicit declaration of function writel; did you mean rv_writel? [-Werror=implicit-function-declaration] writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~~ rv_writel Add <linux/io.h> to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1..b4f68c5 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> +#include <linux/io.h> #include <linux/acpi.h> #include "raven/acp3x.h" -- 2.7.4

3 2

[PATCH] ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout
by Geert Uytterhoeven 05 Mar '20

05 Mar '20

On i386 randconfig: sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume': wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume': wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume': wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset' Fix this by adding the missing dependencies on SND_SOC_AC97_BUS. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- Before commit ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS"), SND_SOC_ALL_CODECS used: select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) but SND_SOC_AC97_BUS_NEW never existed in upstream. Should there be another dependency> See also "non-existent SND_SOC_AC97_BUS_NEW (was: Re: [PATCH v9] ASoC: pxa: switch to new ac97 bus support)" http://lore.kernel.org/r/CAMuHMdU3uxfBwKd8SkOtZSDV5Ai3CKc3CWRhDy0Cz94T1Hn0i… --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e9d54e4576ce5ba..a7e89567edbe8b47 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1610,16 +1610,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW -- 2.17.1

5 6

Re: [PATCH v4 00/10] PCI: Add and use constant PCI_STATUS_ERROR_BITS and helper pci_status_get_and_clear_errors
by David Miller 05 Mar '20

05 Mar '20

Bjorn, please review and let me know if it is OK to merge this via the networking tree. Thank you.

2 2

Problematic motherboard: Supermicro MBD-X11SAT-F-O
by D. R. Evans 04 Mar '20

04 Mar '20

Where is the correct place to post detailed information about a motherboard that provides only unacceptably low volume with ALSA? I realise that the correct place probably isn't here, but I didn't know where else to ask for a pointer to where I should send the detailed information. I initially raised the problem on alsa-users: https://sourceforge.net/p/alsa/mailman/alsa-user/thread/db705c61-87ba-8d66-… but the suggestions made there led to no progress being made, and I am now as sure as I can be that the issue is not user error. The motherboard is a Supermicro MBD-X11SAT-F-O, and the basic high-level device information is: --------- card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 --------- Doc Evans -- Web: http://enginehousebooks.com/drevans

1 0

[PATCH] ASoC: tas571x: use symbolic register offsets
by Hermann Lauer 04 Mar '20

04 Mar '20

convert the numeric register offsets in tas5711_reg_defaults to the defined constants TAS571X_* from tas571x.h Tested with a tas5713 chip for which the driver works, too. Signed-off-by: Hermann Lauer <Hermann.Lauer(a)iwr.uni-heidelberg.de> diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -424,13 +424,13 @@ }; static const struct reg_default tas5711_reg_defaults[] = { - { 0x04, 0x05 }, - { 0x05, 0x40 }, - { 0x06, 0x00 }, - { 0x07, 0xff }, - { 0x08, 0x30 }, - { 0x09, 0x30 }, - { 0x1b, 0x82 }, + { TAS571X_SDI_REG, 0x05 }, + { TAS571X_SYS_CTRL_2_REG, 0x40 }, + { TAS571X_SOFT_MUTE_REG, 0x00 }, + { TAS571X_MVOL_REG, 0xff }, + { TAS571X_CH1_VOL_REG, 0x30 }, + { TAS571X_CH2_VOL_REG, 0x30 }, + { TAS571X_OSC_TRIM_REG, 0x82 }, }; static const struct regmap_config tas5711_regmap_config = {

1 0

Re: [PATCH v4 09/10] PCI: pci-bridge-emul: Use new constant PCI_STATUS_ERROR_BITS
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:28:18PM +0100, Heiner Kallweit wrote: > Use new constant PCI_STATUS_ERROR_BITS to simplify the code. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > --- > drivers/pci/pci-bridge-emul.c | 14 ++------------ > 1 file changed, 2 insertions(+), 12 deletions(-) > > diff --git a/drivers/pci/pci-bridge-emul.c b/drivers/pci/pci-bridge-emul.c > index fffa77093..4f4f54bc7 100644 > --- a/drivers/pci/pci-bridge-emul.c > +++ b/drivers/pci/pci-bridge-emul.c > @@ -50,12 +50,7 @@ static const struct pci_bridge_reg_behavior pci_regs_behavior[] = { > (PCI_STATUS_CAP_LIST | PCI_STATUS_66MHZ | > PCI_STATUS_FAST_BACK | PCI_STATUS_DEVSEL_MASK) << 16), > .rsvd = GENMASK(15, 10) | ((BIT(6) | GENMASK(3, 0)) << 16), > - .w1c = (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) << 16, > + .w1c = PCI_STATUS_ERROR_BITS << 16, > }, > [PCI_CLASS_REVISION / 4] = { .ro = ~0 }, > > @@ -100,12 +95,7 @@ static const struct pci_bridge_reg_behavior pci_regs_behavior[] = { > PCI_STATUS_DEVSEL_MASK) << 16) | > GENMASK(11, 8) | GENMASK(3, 0)), > > - .w1c = (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) << 16, > + .w1c = PCI_STATUS_ERROR_BITS << 16, > > .rsvd = ((BIT(6) | GENMASK(4, 0)) << 16), > }, > -- > 2.25.1 > >

1 0

Re: [PATCH v4 05/10] PCI: Add pci_status_get_and_clear_errors
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:24:23PM +0100, Heiner Kallweit wrote: > Several drivers use the following code sequence: > 1. Read PCI_STATUS > 2. Mask out non-error bits > 3. Action based on error bits set > 4. Write back set error bits to clear them > > As this is a repeated pattern, add a helper to the PCI core. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> Nit: if you have any reason to revise this, I would prefer "pci_status_get_and_clear_errors()" (with parens) in the subject. > --- > v4: > - improve commit description > --- > drivers/pci/pci.c | 23 +++++++++++++++++++++++ > include/linux/pci.h | 1 + > 2 files changed, 24 insertions(+) > > diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c > index d828ca835..c16b0ba2a 100644 > --- a/drivers/pci/pci.c > +++ b/drivers/pci/pci.c > @@ -173,6 +173,29 @@ unsigned char pci_bus_max_busnr(struct pci_bus *bus) > } > EXPORT_SYMBOL_GPL(pci_bus_max_busnr); > > +/** > + * pci_status_get_and_clear_errors - return and clear error bits in PCI_STATUS > + * @pdev: the PCI device > + * > + * Returns error bits set in PCI_STATUS and clears them. > + */ > +int pci_status_get_and_clear_errors(struct pci_dev *pdev) > +{ > + u16 status; > + int ret; > + > + ret = pci_read_config_word(pdev, PCI_STATUS, &status); > + if (ret != PCIBIOS_SUCCESSFUL) > + return -EIO; > + > + status &= PCI_STATUS_ERROR_BITS; > + if (status) > + pci_write_config_word(pdev, PCI_STATUS, status); > + > + return status; > +} > +EXPORT_SYMBOL_GPL(pci_status_get_and_clear_errors); > + > #ifdef CONFIG_HAS_IOMEM > void __iomem *pci_ioremap_bar(struct pci_dev *pdev, int bar) > { > diff --git a/include/linux/pci.h b/include/linux/pci.h > index 101d71e0a..7beaf51e9 100644 > --- a/include/linux/pci.h > +++ b/include/linux/pci.h > @@ -1210,6 +1210,7 @@ int pci_select_bars(struct pci_dev *dev, unsigned long flags); > bool pci_device_is_present(struct pci_dev *pdev); > void pci_ignore_hotplug(struct pci_dev *dev); > struct pci_dev *pci_real_dma_dev(struct pci_dev *dev); > +int pci_status_get_and_clear_errors(struct pci_dev *pdev); > > int __printf(6, 7) pci_request_irq(struct pci_dev *dev, unsigned int nr, > irq_handler_t handler, irq_handler_t thread_fn, void *dev_id, > -- > 2.25.1 > >

1 0

Re: [PATCH v4 04/10] PCI: Add constant PCI_STATUS_ERROR_BITS
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:23:44PM +0100, Heiner Kallweit wrote: > This collection of PCI error bits is used in more than one driver, > so move it to the PCI core. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > --- > v4: > - move new constant to include/linux/pci.h > - improve commit description > --- > drivers/net/ethernet/marvell/skge.h | 7 ------- > drivers/net/ethernet/marvell/sky2.h | 7 ------- > include/linux/pci.h | 7 +++++++ > 3 files changed, 7 insertions(+), 14 deletions(-) > > diff --git a/drivers/net/ethernet/marvell/skge.h b/drivers/net/ethernet/marvell/skge.h > index e76c03c87..e149bdfe1 100644 > --- a/drivers/net/ethernet/marvell/skge.h > +++ b/drivers/net/ethernet/marvell/skge.h > @@ -15,13 +15,6 @@ > #define PCI_VPD_ROM_SZ 7L<<14 /* VPD ROM size 0=256, 1=512, ... */ > #define PCI_REV_DESC 1<<2 /* Reverse Descriptor bytes */ > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_SIG_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/drivers/net/ethernet/marvell/sky2.h b/drivers/net/ethernet/marvell/sky2.h > index aee87f838..851d8ed34 100644 > --- a/drivers/net/ethernet/marvell/sky2.h > +++ b/drivers/net/ethernet/marvell/sky2.h > @@ -252,13 +252,6 @@ enum { > }; > > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_SIG_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/include/linux/pci.h b/include/linux/pci.h > index 3840a541a..101d71e0a 100644 > --- a/include/linux/pci.h > +++ b/include/linux/pci.h > @@ -42,6 +42,13 @@ > > #include <linux/pci_ids.h> > > +#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > + PCI_STATUS_SIG_SYSTEM_ERROR | \ > + PCI_STATUS_REC_MASTER_ABORT | \ > + PCI_STATUS_REC_TARGET_ABORT | \ > + PCI_STATUS_SIG_TARGET_ABORT | \ > + PCI_STATUS_PARITY) > + > /* > * The PCI interface treats multi-function devices as independent > * devices. The slot/function address of each device is encoded > -- > 2.25.1 > >

1 0

Applied "ASoC: stm32: sai: manage rebind issue" to the asoc tree
by Mark Brown 04 Mar '20

04 Mar '20

The patch ASoC: stm32: sai: manage rebind issue has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0d6defc7e0e437a9fd53622f7fd85740f38d5693 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Wed, 4 Mar 2020 11:24:06 +0100 Subject: [PATCH] ASoC: stm32: sai: manage rebind issue The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a..10eb4b8e8e7e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH] ASoC: stm32: sai: manage rebind issue
by Olivier Moysan 04 Mar '20

04 Mar '20

The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0bbf9ed5e48b..48dfcdd13574 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,21 +1547,21 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); - if (ret) - return ret; - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; } + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); + if (ret) + return ret; + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + return 0; } @@ -1570,6 +1570,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.17.1

1 0

Build is failing with python 3.8
by GitHub issues - opened 04 Mar '20

04 Mar '20

alsa-project/alsa-lib issue #33 was opened from seb128: One build log from Ubuntu (1.1.9 but the current version has the same issue) https://launchpadlibrarian.net/462093809/buildlog_ubuntu-focal-amd64.alsa-l… The Ubuntu package has been fixed with that change http://launchpadlibrarian.net/462101751/alsa-lib_1.2.1.2-2_1.2.1.2-2ubuntu1… The issue/needed changed is explained on https://docs.python.org/3/whatsnew/3.8.html > To embed Python into an application, a new --embed option must be passed to python3-config --libs --embed to get -lpython3.8 (link the application to libpython). To support both 3.8 and older, try python3-config --libs --embed first and fallback to python3-config --libs (without --embed) if the previous command fails. The Ubuntu patch doesn't have that fallback logic though which is why I'm not proposing it with a merge request Issue URL : https://github.com/alsa-project/alsa-lib/issues/33 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: SOF: Fix snd_sof_ipc_stream_posn()
by Dan Carpenter 04 Mar '20

04 Mar '20

We're passing "&posn" instead of "posn" so it ends up corrupting memory instead of doing something useful. Fixes: 53e0c72d98ba ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 22d296f95761..d6bf53dcd80b 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -524,7 +524,7 @@ int snd_sof_ipc_stream_posn(struct snd_soc_component *scomp, /* send IPC to the DSP */ err = sof_ipc_tx_message(sdev->ipc, - stream.hdr.cmd, &stream, sizeof(stream), &posn, + stream.hdr.cmd, &stream, sizeof(stream), posn, sizeof(*posn)); if (err < 0) { dev_err(sdev->dev, "error: failed to get stream %d position\n", -- 2.11.0

6 5

[PATCH] ASoC: soc-pcm: Revert "call snd_soc_component_open/close() once"
by Kai Vehmanen 04 Mar '20

04 Mar '20

ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic in the reverted patch was not sufficient and led to PCM playback and module unload errors. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 19 +++++++++++++------ 3 files changed, 22 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e7..14e175cdeeb8 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..0c3a12d6290a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -466,13 +466,16 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } -static int soc_pcm_components_open(struct snd_pcm_substream *substream) +static int soc_pcm_components_open(struct snd_pcm_substream *substream, + struct snd_soc_component **last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + *last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, @@ -489,17 +492,21 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream) return ret; } } - + *last = NULL; return 0; } -static int soc_pcm_components_close(struct snd_pcm_substream *substream) +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + r = snd_soc_component_close(component, substream); if (r < 0) ret = r; /* use last ret */ @@ -536,7 +543,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, NULL); snd_soc_dapm_stream_stop(rtd, substream->stream); @@ -577,7 +584,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - ret = soc_pcm_components_open(substream); + ret = soc_pcm_components_open(substream, &component); if (ret < 0) goto component_err; @@ -682,7 +689,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); component_err: - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, component); mutex_unlock(&rtd->card->pcm_mutex); -- 2.17.1

6 26

[PATCH] ASoC: rt1015: modify pre-divider for sysclk
by jack.yu＠realtek.com 03 Mar '20

03 Mar '20

From: Jack Yu <jack.yu(a)realtek.com> Modify pre-divider for system clock. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 5fa21d8ab620..bb310bc7febd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -664,7 +664,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, RT1015_TDM_MASTER, RT1015_I2S_DL_MASK, val_len); snd_soc_component_update_bits(component, RT1015_CLK2, - RT1015_FS_PD_MASK, pre_div); + RT1015_FS_PD_MASK, pre_div << RT1015_FS_PD_SFT); return 0; } -- 2.25.1

2 2

[PATCH] ASoC: amd: AMD RV RT5682 should depends on CROS_EC
by Enric Balletbo i Serra 03 Mar '20

03 Mar '20

If SND_SOC_AMD_RV_RT5682_MACH=y, below kconfig and build errors can be seen: WARNING: unmet direct dependencies detected for SND_SOC_CROS_EC_CODEC WARNING: unmet direct dependencies detected for I2C_CROS_EC_TUNNEL ld: drivers/i2c/busses/i2c-cros-ec-tunnel.o: in function `ec_i2c_xfer': i2c-cros-ec-tunnel.c:(.text+0x2fc): undefined reference to `cros_ec_cmd_xfer_status' ld: sound/soc/codecs/cros_ec_codec.o: in function `wov_host_event': cros_ec_codec.c:(.text+0x4fb): undefined reference to `cros_ec_get_host_event' ld: sound/soc/codecs/cros_ec_codec.o: in function `send_ec_host_command': cros_ec_codec.c:(.text+0x831): undefined reference to `cros_ec_cmd_xfer_status' This is because it will select SND_SOC_CROS_EC_CODEC and I2c_CROS_EC_TUNNEL but both depends on CROS_EC. Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index b29ef1373946..bce4cee5cb54 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -33,6 +33,6 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL - depends on SND_SOC_AMD_ACP3x && I2C + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. -- 2.25.1

3 2

Applied "ASoC: amd: Fix compile warning of argument type" to the asoc tree
by Mark Brown 03 Mar '20

03 Mar '20

The patch ASoC: amd: Fix compile warning of argument type has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d7729c40b376ab7c1ad4b0498a8bbf44bed3cbf4 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Tue, 3 Mar 2020 14:34:37 +0530 Subject: [PATCH] ASoC: amd: Fix compile warning of argument type Fixes: >> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' >> expects argument of type 'int', but argument 3 has type 'long int' >> [-Wformat=] dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Link: https://lore.kernel.org/r/20200303090444.95805-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.20.1

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[PATCH v3 0/3] simple-audio-card codec2codec support
by Samuel Holland 03 Mar '20

03 Mar '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when all DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Changes in v3: - Update use of for_each_rtd_components for v5.6 Changes in v2: - Drop patch 1 as it was merged - Automatically detect codec2codec links instead of using a DT property Samuel Holland (3): ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 49 +++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 115 insertions(+), 28 deletions(-) -- 2.24.1

2 6

[RFC PATCH 0/2] Introduce multi PM domains helpers
by Daniel Baluta 03 Mar '20

03 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 54 +++------------------ 3 files changed, 118 insertions(+), 48 deletions(-) -- 2.17.1

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[PATCH][next] ASoc: amd: use correct format specifier for a long int
by Colin King 03 Mar '20

03 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> Currently the format specifier %d is being used for a long int. Fix the by using %ld instead. Fixes warning: sound/soc/amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat=] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.25.0

2 1

Change PCM device number of Asus Xonar U5
by GitHub pull_request - opened 03 Mar '20

03 Mar '20

alsa-project/alsa-lib pull request #32 was opened from be-marc: Asus Xonar U5 does not use the first PCM device for digital data like the U7. Request URL : https://github.com/alsa-project/alsa-lib/pull/32 Patch URL : https://github.com/alsa-project/alsa-lib/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: amd: Fix compile warning of argument type
by Akshu Agrawal 03 Mar '20

03 Mar '20

Fixes: >> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' >> expects argument of type 'int', but argument 3 has type 'long int' >> [-Wformat=] dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reported-by: kbuild test robot <lkp(a)intel.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.17.1

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[PATCH v2] soundwire: bus: provide correct return value on error
by Pierre-Louis Bossart 03 Mar '20

03 Mar '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> It seems to be a typo. It makes more sense to return the return value of sdw_update() instead of the value we want to update. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: address Vinod's comment and use single return drivers/soundwire/bus.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index ccaa590df61e..488c3c9e4947 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1056,13 +1056,10 @@ static int sdw_initialize_slave(struct sdw_slave *slave) val |= SDW_DP0_INT_PORT_READY | SDW_DP0_INT_BRA_FAILURE; ret = sdw_update(slave, SDW_DP0_INTMASK, val, val); - if (ret < 0) { + if (ret < 0) dev_err(slave->bus->dev, "SDW_DP0_INTMASK read failed:%d\n", ret); - return val; - } - - return 0; + return ret; } static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) -- 2.20.1

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[PATCH AUTOSEL 4.19 01/32] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 60cb00fd0c693..84b44cdae28a1 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -262,6 +262,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -283,9 +284,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 01/58] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cfab60d88c921..09ff209df4a30 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -254,6 +254,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -280,9 +281,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 01/66] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cfab60d88c921..09ff209df4a30 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -254,6 +254,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -280,9 +281,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH 0/9] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 76 ++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 267 insertions(+), 69 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

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[PATCH] ASoC: rt1015: add operation callback function for rt1015_dai[]
by jack.yu＠realtek.com 02 Mar '20

02 Mar '20

From: Jack Yu <jack.yu(a)realtek.com> Add operation callback function for rt1015_dai[]. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c118d030bd2d..5fa21d8ab620 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.25.1

2 1

[PATCH 0/4] ASoC: SOF: updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

4 unrelated improvements grouped in one bundle. Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp include/sound/sof/topology.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/intel/hda-loader.c | 6 ++++++ sound/soc/sof/ipc.c | 12 +++++++----- sound/soc/sof/pcm.c | 6 ++---- 5 files changed, 18 insertions(+), 11 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

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