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[PATCH 1/4] ASoC: cs42l42: Fix 1536000 Bit Clock instability
by Lucas Tanure 25 May '21

25 May '21

The 16 Bits, 2 channels, 48K sample rate use case needs to configure a safer pll_divout during the start of PLL After 800us from the start of PLL the correct pll_divout can be set Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 47 +++++++++++++++++++++++++------------- sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 33 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index b084acd1e86b..94788a55fa3b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -589,6 +589,7 @@ struct cs42l42_pll_params { u8 pll_divout; u32 mclk_int; u8 pll_cal_ratio; + u8 n; }; /* @@ -596,21 +597,21 @@ struct cs42l42_pll_params { * Table 4-5 from the Datasheet */ static const struct cs42l42_pll_params pll_ratio_table[] = { - { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125 }, - { 2822400, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128 }, - { 3000000, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128 }, - { 3072000, 0, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125 }, - { 4000000, 0, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96 }, - { 4096000, 0, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94 }, - { 5644800, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128 }, - { 6000000, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128 }, - { 6144000, 0, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125 }, - { 11289600, 0, 0, 0, 0, 0, 0, 0, 11289600, 0 }, - { 12000000, 0, 0, 0, 0, 0, 0, 0, 12000000, 0 }, - { 12288000, 0, 0, 0, 0, 0, 0, 0, 12288000, 0 }, - { 22579200, 1, 0, 0, 0, 0, 0, 0, 22579200, 0 }, - { 24000000, 1, 0, 0, 0, 0, 0, 0, 24000000, 0 }, - { 24576000, 1, 0, 0, 0, 0, 0, 0, 24576000, 0 } + { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2}, + { 2822400, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 3000000, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 3072000, 0, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 4000000, 0, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96, 1}, + { 4096000, 0, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94, 1}, + { 5644800, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 6000000, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 6144000, 0, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 11289600, 0, 0, 0, 0, 0, 0, 0, 11289600, 0, 1}, + { 12000000, 0, 0, 0, 0, 0, 0, 0, 12000000, 0, 1}, + { 12288000, 0, 0, 0, 0, 0, 0, 0, 12288000, 0, 1}, + { 22579200, 1, 0, 0, 0, 0, 0, 0, 22579200, 0, 1}, + { 24000000, 1, 0, 0, 0, 0, 0, 0, 24000000, 0, 1}, + { 24576000, 1, 0, 0, 0, 0, 0, 0, 24576000, 0, 1} }; static int cs42l42_pll_config(struct snd_soc_component *component) @@ -746,8 +747,12 @@ static int cs42l42_pll_config(struct snd_soc_component *component) snd_soc_component_update_bits(component, CS42L42_PLL_CTL3, CS42L42_PLL_DIVOUT_MASK, - pll_ratio_table[i].pll_divout + (pll_ratio_table[i].pll_divout * pll_ratio_table[i].n) << CS42L42_PLL_DIVOUT_SHIFT); + if (pll_ratio_table[i].n != 1) + cs42l42->pll_divout = pll_ratio_table[i].pll_divout; + else + cs42l42->pll_divout = 0; snd_soc_component_update_bits(component, CS42L42_PLL_CAL_RATIO, CS42L42_PLL_CAL_RATIO_MASK, @@ -902,6 +907,16 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); + + if (cs42l42->pll_divout) { + usleep_range(CS42L42_PLL_DIVOUT_TIME_US, + CS42L42_PLL_DIVOUT_TIME_US * 2); + snd_soc_component_update_bits(component, CS42L42_PLL_CTL3, + CS42L42_PLL_DIVOUT_MASK, + cs42l42->pll_divout << + CS42L42_PLL_DIVOUT_SHIFT); + } + ret = regmap_read_poll_timeout(cs42l42->regmap, CS42L42_PLL_LOCK_STATUS, regval, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 24f7be228d5f..7bf05ff05f74 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -755,6 +755,7 @@ #define CS42L42_NUM_SUPPLIES 5 #define CS42L42_BOOT_TIME_US 3000 +#define CS42L42_PLL_DIVOUT_TIME_US 800 #define CS42L42_CLOCK_SWITCH_DELAY_US 150 #define CS42L42_PLL_LOCK_POLL_US 250 #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 @@ -777,6 +778,7 @@ struct cs42l42_private { int bclk; u32 sclk; u32 srate; + u8 pll_divout; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.31.1

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[PATCH v2] ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
by Peter Ujfalusi 25 May '21

25 May '21

Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces bug plock/pop noise over line out (green jack on the back) which can be fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine. Convert the existing entry for the same SSID to chain to apply this fixup as well. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, Changes since v1 - instead of deny_list use chained fixup to fix the plock/pop noise Changes since RFC: - Fixed up the PCI ID ever since I have this workstation (three weeks) the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 Missing details from alsa-info output: # lspci -nnk | grep -A2 Audio 00:1f.3 Audio device [0403]: Intel Corporation Comet Lake PCH cAVS [8086:06c8] DeviceName: Onboard - Sound Subsystem: Lenovo Device [17aa:1048] CPU: i9-10900k The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? [1] https://lore.kernel.org/alsa-devel/20210524152533.7479-1-peter.ujfalusi@lin… Thanks you, Peter Ujfalusi sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3e269de84079..e2e0ee7b137d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6543,6 +6543,7 @@ enum { ALC295_FIXUP_ASUS_DACS, ALC295_FIXUP_HP_OMEN, ALC285_FIXUP_HP_SPECTRE_X360, + ALC623_FIXUP_LENOVO_THINKSTATION_P340, }; static const struct hda_fixup alc269_fixups[] = { @@ -8109,6 +8110,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, }, + [ALC623_FIXUP_LENOVO_THINKSTATION_P340] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC283_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8412,7 +8419,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), - SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -8677,6 +8684,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.31.1

3 2

[PATCH] ALSA: hda: Add Lenovo ThinkStation P340 to power_save_denylist
by Peter Ujfalusi 25 May '21

25 May '21

On playback start there is a huge plock/pop noise via the green jack used in Line out mode. The only way I was able to fix it to set the power_save to 0 for the snd_hda_intel. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, Changes since RFC: - Fixed up the PCI ID ever since I have this workstation (three weeks) the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 Missing details from alsa-info output: # lspci -nnk | grep -A2 Audio 00:1f.3 Audio device [0403]: Intel Corporation Comet Lake PCH cAVS [8086:06c8] DeviceName: Onboard - Sound Subsystem: Lenovo Device [17aa:1048] CPU: i9-10900k The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? Thanks you, Peter Ujfalusi sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..9cacd1cb211b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2213,6 +2213,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Loud plock/pop noise via the green jack on playback start */ + SND_PCI_QUIRK(0x17aa, 0x1048, "Lenovo ThinkStation P340", 0), {} }; #endif /* CONFIG_PM */ -- 2.31.1

3 6

[PATCH] ASoC: Intel: boards: Select SND_SOC_HDAC_HDMI for Soundwire machine driver
by Richard Fitzgerald 25 May '21

25 May '21

Add select of SND_SOC_HDAC_HDMI for SND_SOC_INTEL_SOUNDWIRE_SOF_MACH. Without this the build results in missing references to hdac_hdmi_jack_port_init and hdac_hdmi_jack_init. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index eef5f4ac87c5..6a1639b14b67 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -601,6 +601,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON + select SND_SOC_HDAC_HDMI help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 -- 2.20.1

3 3

[PATCH 00/11] drm/vc4: hdmi: Enable Channel Mapping, IEC958, HBR Passthrough using hdmi-codec
by Maxime Ripard 25 May '21

25 May '21

Hi, hdmi-codec allows to have a lot of HDMI-audio related infrastructure in place, it's missing a few controls to be able to provide HBR passthrough. This series adds more infrastructure for the drivers, and leverages it in the vc4 HDMI controller driver. One thing that felt a bit weird is that even though https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-dri… mentions that the iec958 mask control should be a mixer control and the default control should be a PCM one, it feels a bit weird to have two different control type for two controls so similar, and other drivers are pretty inconsistent with this. Should we update the documentation? Thanks! Maxime Dom Cobley (5): drm/vc4: hdmi: Set HD_CTL_WHOLSMP and HD_CTL_CHALIGN_SET drm/vc4: hdmi: Set HDMI_MAI_FMT drm/vc4: hdmi: Set VC4_HDMI_MAI_CONFIG_FORMAT_REVERSE drm/vc4: hdmi: Remove firmware logic for MAI threshold setting ARM: dts: bcm2711: Tune DMA parameters for HDMI audio Maxime Ripard (6): snd: iec958: split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook drm/vc4: hdmi: Register HDMI codec drm/vc4: hdmi: Remove redundant variables arch/arm/boot/dts/bcm2711.dtsi | 4 +- drivers/gpu/drm/vc4/Kconfig | 1 + drivers/gpu/drm/vc4/vc4_hdmi.c | 322 ++++++++++++++------------------- drivers/gpu/drm/vc4/vc4_hdmi.h | 5 +- drivers/gpu/drm/vc4/vc4_regs.h | 30 +++ include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 131 +++++++++----- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++----- 9 files changed, 456 insertions(+), 276 deletions(-) -- 2.31.1

2 17

[PATCH -next] ALSA: control_led - use DEVICE_ATTR_*() macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR, which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/control_led.c | 34 +++++++++++++++++++--------------- 1 file changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a5e751f26d46 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -375,7 +375,7 @@ static void snd_ctl_led_disconnect(struct snd_card *card) * sysfs */ -static ssize_t show_mode(struct device *dev, +static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -390,7 +390,8 @@ static ssize_t show_mode(struct device *dev, return sprintf(buf, "%s\n", str); } -static ssize_t store_mode(struct device *dev, struct device_attribute *attr, +static ssize_t mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -419,7 +420,7 @@ static ssize_t store_mode(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t show_brightness(struct device *dev, +static ssize_t brightness_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -427,8 +428,8 @@ static ssize_t show_brightness(struct device *dev, return sprintf(buf, "%u\n", ledtrig_audio_get(led->trigger_type)); } -static DEVICE_ATTR(mode, 0644, show_mode, store_mode); -static DEVICE_ATTR(brightness, 0444, show_brightness, NULL); +static DEVICE_ATTR_RW(mode); +static DEVICE_ATTR_RO(brightness); static struct attribute *snd_ctl_led_dev_attrs[] = { &dev_attr_mode.attr, @@ -562,22 +563,25 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si return count; } -static ssize_t parse_attach(struct device *dev, struct device_attribute *attr, +static ssize_t attach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, true); } -static ssize_t parse_detach(struct device *dev, struct device_attribute *attr, +static ssize_t detach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, false); } -static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t reset_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); int err; @@ -590,8 +594,8 @@ static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t ctl_list(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t list_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); struct snd_card *card; @@ -624,10 +628,10 @@ static ssize_t ctl_list(struct device *dev, return buf2 - buf; } -static DEVICE_ATTR(attach, 0200, NULL, parse_attach); -static DEVICE_ATTR(detach, 0200, NULL, parse_detach); -static DEVICE_ATTR(reset, 0200, NULL, ctl_reset); -static DEVICE_ATTR(list, 0444, ctl_list, NULL); +static DEVICE_ATTR_WO(attach); +static DEVICE_ATTR_WO(detach); +static DEVICE_ATTR_WO(reset); +static DEVICE_ATTR_RO(list); static struct attribute *snd_ctl_led_card_attrs[] = { &dev_attr_attach.attr, -- 2.17.1

3 2

[PATCH] ALSA: line6: Improve poor error handling in line6_init_cap_control
by Hyeonggon Yoo 25 May '21

25 May '21

line6_init_cap_control doesn't free resources properly when allocations like kmalloc, usb_alloc_urb fails. It can cause memory leak when some allocations succeed, and then an error occurs. This patch makes line6_init_cap_control to properly free the resources, freeing previously allocated resources when there's an error. Signed-off-by: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> --- sound/usb/line6/driver.c | 36 +++++++++++++++++++++++++++--------- 1 file changed, 27 insertions(+), 9 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 9602929b7de9..6991cb855023 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -688,34 +688,52 @@ static int line6_init_cap_control(struct usb_line6 *line6) /* initialize USB buffers: */ line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); - if (!line6->buffer_listen) - return -ENOMEM; + if (!line6->buffer_listen) { + ret = -ENOMEM; + goto fail_ret; + } line6->urb_listen = usb_alloc_urb(0, GFP_KERNEL); - if (!line6->urb_listen) - return -ENOMEM; + if (!line6->urb_listen) { + ret = -ENOMEM; + goto fail1; + } if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); - if (!line6->buffer_message) - return -ENOMEM; + if (!line6->buffer_message) { + ret = -ENOMEM; + goto fail2; + } ret = line6_init_midi(line6); if (ret < 0) - return ret; + goto fail3; } else { ret = line6_hwdep_init(line6); if (ret < 0) - return ret; + goto fail2; } ret = line6_start_listen(line6); if (ret < 0) { dev_err(line6->ifcdev, "cannot start listening: %d\n", ret); - return ret; + if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) + goto fail3; + else + goto fail2; } return 0; + +fail3: + kfree(line6->buffer_message); +fail2: + usb_free_urb(line6->urb_listen); +fail1: + kfree(line6->buffer_listen); +fail_ret: + return ret; } static void line6_startup_work(struct work_struct *work) -- 2.25.1

2 3

[GIT PULL] ASoC fixes for v5.13-rc3
by Mark Brown 25 May '21

25 May '21

The following changes since commit 25c4a9b614f101bb9f3e687960815db7dc439c0f: ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2 (2021-04-23 18:13:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc3 for you to fetch changes up to af2702549d68519ac78228e915d9b2c199056787: ASoC: qcom: lpass-cpu: Use optional clk APIs (2021-05-21 13:12:29 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A collection of fixes that have come in since the merge window, mainly device specific things. The fixes to the generic cards from Morimoto-san are handling regressions that were introduced in the merge window on at least the Kontron sl28-var3-ads2. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency Bixuan Cui (1): ASoC: codecs: lpass-tx-macro: add missing MODULE_DEVICE_TABLE Charles Keepax (5): ASoC: cs53l30: Add missing regmap use_single config ASoC: cs42l73: Add missing regmap use_single config ASoC: cs35l34: Add missing regmap use_single config ASoC: cs35l32: Add missing regmap use_single config ASoC: cs42l52: Minor tidy up of error paths Dan Carpenter (1): ASoC: cs35l33: fix an error code in probe() Hans de Goede (2): ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet Jerome Brunet (2): ASoC: stm32: do not request a new clock consummer reference ASoC: da7219: do not request a new clock consummer reference Kuninori Morimoto (5): ASoC: simple-card: add simple_parse_node() ASoC: simple-card: add simple_link_init() ASoC: audio-graph: tidyup graph_dai_link_of_dpcm() ASoC: audio-graph: tidyup graph_parse_node() ASoC: soc-dai.h: Align the word of comment for SND_SOC_DAIFMT_CBC_CFC Marco Felsch (1): ASoC: max98088: fix ni clock divider calculation Mark Brown (2): Merge series "ASoC: Revert clk_hw_get_clk() cleanup" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: simple-card / audio-graph re-cleanup" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Ranjani Sridharan (1): ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware Richard Fitzgerald (1): ASoC: cs42l42: Regmap must use_single_read/write Shengjiu Wang (1): ASoC: ak5558: Correct the dai name for ak5552 Shuming Fan (1): ASoC: rt711-sdca: fix the function number of SDCA control for feature unit 0x1E Srinivas Kandagatla (1): ASoC: codecs: lpass-rx-macro: add missing MODULE_DEVICE_TABLE Stephen Boyd (1): ASoC: qcom: lpass-cpu: Use optional clk APIs Vijendar Mukunda (1): ASoC: amd: fix for pcm_read() error Zou Wei (1): ASoC: sti-sas: add missing MODULE_DEVICE_TABLE include/sound/soc-dai.h | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 -- sound/soc/amd/raven/acp3x.h | 1 + sound/soc/amd/raven/pci-acp3x.c | 15 +++ sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l32.c | 3 + sound/soc/codecs/cs35l33.c | 1 + sound/soc/codecs/cs35l34.c | 3 + sound/soc/codecs/cs42l42.c | 3 + sound/soc/codecs/cs42l56.c | 7 +- sound/soc/codecs/cs42l73.c | 3 + sound/soc/codecs/cs53l30.c | 3 + sound/soc/codecs/da7219.c | 5 +- sound/soc/codecs/lpass-rx-macro.c | 1 + sound/soc/codecs/lpass-tx-macro.c | 1 + sound/soc/codecs/max98088.c | 13 ++- sound/soc/codecs/rt711-sdca.c | 4 +- sound/soc/codecs/sti-sas.c | 1 + sound/soc/fsl/Kconfig | 1 + sound/soc/generic/audio-graph-card.c | 57 ++++++------ sound/soc/generic/simple-card.c | 168 +++++++++++++++++----------------- sound/soc/intel/boards/bytcr_rt5640.c | 25 +++++ sound/soc/qcom/lpass-cpu.c | 12 +-- sound/soc/sof/intel/hda-dai.c | 5 + sound/soc/stm/stm32_sai_sub.c | 5 +- 25 files changed, 202 insertions(+), 149 deletions(-)

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[PATCH] ASoC: meson: use dev_err_probe
by Jerome Brunet 25 May '21

25 May '21

Use dev_err_probe() helper function to handle probe deferral. It removes the open coded test for -EPROBE_DEFER but more importantly, it sets the deferral reason in debugfs which is great for debugging. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu.c | 17 +++++++---------- sound/soc/meson/axg-fifo.c | 11 +++++------ sound/soc/meson/axg-pdm.c | 9 +++------ sound/soc/meson/axg-spdifin.c | 6 ++---- sound/soc/meson/axg-spdifout.c | 6 ++---- sound/soc/meson/axg-tdm-formatter.c | 18 ++++++------------ sound/soc/meson/axg-tdm-interface.c | 10 ++++------ sound/soc/meson/meson-card-utils.c | 4 ++-- sound/soc/meson/t9015.c | 7 +++---- 9 files changed, 34 insertions(+), 54 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index ba15d5762b0b..37036adf14ce 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -219,31 +219,29 @@ static int aiu_clk_get(struct device *dev) aiu->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(aiu->pclk)) { - if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu pclk\n"); + dev_err_probe(dev, PTR_ERR(aiu->pclk), + "Can't get the aiu pclk\n"); return PTR_ERR(aiu->pclk); } aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) { - if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu spdif master clock\n"); + dev_err_probe(dev, PTR_ERR(aiu->spdif_mclk), + "Can't get the aiu spdif master clock\n"); return PTR_ERR(aiu->spdif_mclk); } ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), &aiu->i2s); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the i2s clocks\n"); + dev_err_probe(dev, ret, "Can't get the i2s clocks\n"); return ret; } ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), &aiu->spdif); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the spdif clocks\n"); + dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); return ret; } @@ -282,8 +280,7 @@ static int aiu_probe(struct platform_device *pdev) ret = device_reset(dev); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Failed to reset device\n"); + dev_err_probe(dev, ret, "Failed to reset device\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b9af2d513e09..d670a32d56ab 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -352,17 +352,16 @@ int axg_fifo_probe(struct platform_device *pdev) fifo->pclk = devm_clk_get(dev, NULL); if (IS_ERR(fifo->pclk)) { - if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %ld\n", - PTR_ERR(fifo->pclk)); + dev_err_probe(dev, PTR_ERR(fifo->pclk), + "failed to get pclk: %ld\n", PTR_ERR(fifo->pclk)); return PTR_ERR(fifo->pclk); } fifo->arb = devm_reset_control_get_exclusive(dev, NULL); if (IS_ERR(fifo->arb)) { - if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) - dev_err(dev, "failed to get arb reset: %ld\n", - PTR_ERR(fifo->arb)); + dev_err_probe(dev, PTR_ERR(fifo->arb), + "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); return PTR_ERR(fifo->arb); } diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index bfd37d49a73e..f630311b5b29 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -613,24 +613,21 @@ static int axg_pdm_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->dclk = devm_clk_get(dev, "dclk"); if (IS_ERR(priv->dclk)) { ret = PTR_ERR(priv->dclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } priv->sysclk = devm_clk_get(dev, "sysclk"); if (IS_ERR(priv->sysclk)) { ret = PTR_ERR(priv->sysclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..8f587c3bb5d7 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -481,16 +481,14 @@ static int axg_spdifin_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->refclk = devm_clk_get(dev, "refclk"); if (IS_ERR(priv->refclk)) { ret = PTR_ERR(priv->refclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index e769a5ee6e27..58e02f6898b7 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -424,16 +424,14 @@ static int axg_spdifout_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index cab7fa2851aa..108f11e14006 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -284,8 +284,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(formatter->pclk)) { ret = PTR_ERR(formatter->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } @@ -293,8 +292,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(formatter->sclk)) { ret = PTR_ERR(formatter->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -302,8 +300,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(formatter->lrclk)) { ret = PTR_ERR(formatter->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -311,8 +308,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); if (IS_ERR(formatter->sclk_sel)) { ret = PTR_ERR(formatter->sclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk_sel: %d\n", ret); return ret; } @@ -320,8 +316,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); if (IS_ERR(formatter->lrclk_sel)) { ret = PTR_ERR(formatter->lrclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk_sel: %d\n", ret); return ret; } @@ -329,8 +324,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get reset: %d\n", ret); + dev_err_probe(dev, ret, "failed to get reset: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 87cac440b369..a2c7f658f6b4 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -519,8 +519,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(iface->sclk)) { ret = PTR_ERR(iface->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -528,8 +527,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(iface->lrclk)) { ret = PTR_ERR(iface->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -545,8 +543,8 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) if (ret == -ENOENT) { iface->mclk = NULL; } else { - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", + ret); return ret; } } diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 300ac8be46ef..ac062c03cf33 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -86,8 +86,8 @@ int meson_card_parse_dai(struct snd_soc_card *card, ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); + dev_err_probe(card->dev, ret, + "can't parse sound-dai at %pOFn\n", node); return ret; } *dai_of_node = args.np; diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 4c1349dd1e06..7ff16f51dbd4 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -259,15 +259,14 @@ static int t9015_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { - if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get core clock\n"); + dev_err_probe(dev, PTR_ERR(priv->pclk), + "failed to get core clock\n"); return PTR_ERR(priv->pclk); } priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) { - if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) - dev_err(dev, "failed to AVDD\n"); + dev_err_probe(dev, PTR_ERR(priv->avdd), "failed to AVDD\n"); return PTR_ERR(priv->avdd); } -- 2.32.0.rc0

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[PATCH -next] ALSA: pcm: use DEVICE_ATTR_RO macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index b163164a83ec..6fd3677685d7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1004,7 +1004,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->pstr->substream_opened--; } -static ssize_t show_pcm_class(struct device *dev, +static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); @@ -1024,7 +1024,7 @@ static ssize_t show_pcm_class(struct device *dev, return sprintf(buf, "%s\n", str); } -static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); +static DEVICE_ATTR_RO(pcm_class); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL -- 2.17.1

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