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

wallclock time for midi events without audio port
by Michael Kirchner 17 Dec '20

17 Dec '20

Hi, i try to get an absolute wallclock time for midi events using event->time.time.tv_sec. As I understand I have to add trigger time to get wall clock time. Therefor I use snd_pcm_status_get_trigger_htstamp(). If I use alsa audio via snd_pcm_open there is no problem. But: how can I get trigger time if I use only a midi port without audio? (usecase is: soft synth with alsa midi in -> jack audio out) Sincerely, Michael

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utils/alsa.m4: update for newer autoconf.
by GitHub pull_request - opened 17 Dec '20

17 Dec '20

alsa-project/alsa-lib pull request #106 was opened from sezero: fixes autoconf-2.70 warnings. should be compatible down to autoconf-2.59. Request URL : https://github.com/alsa-project/alsa-lib/pull/106 Patch URL : https://github.com/alsa-project/alsa-lib/pull/106.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH alsa-lib 1/2] pcm: Add snd_pcm_audio_tstamp_type_t constants
by Takashi Iwai 17 Dec '20

17 Dec '20

From: David Henningsson <diwic(a)ubuntu.com> These are mostly a copy-paste from the kernel headers. But since functions snd_pcm_audio_tstamp_config make use of these they should be added to the public API as well. Reported-by: Alex Moon <alex.r.moon(a)gmail.com> Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/pcm.h b/include/pcm.h index 5b0782315585..cf1eea8bbc9b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -350,6 +350,20 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + /** + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /**< DMA time, reported as per hw_ptr */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /**< link time reported by sample or wallclock counter, reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +} snd_pcm_audio_tstamp_type_t; + typedef struct _snd_pcm_audio_tstamp_config { /* 5 of max 16 bits used */ unsigned int type_requested:4; -- 2.26.2

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Re: [PATCH 01/01] Add VID to support native DSD reproduction on FiiO devices.
by Amadej Kastelic 17 Dec '20

17 Dec '20

Message-Id: <c759bc50134ea434bfed0a183e0ce72984f5b609.camel(a)gmail.com> Add VID to support native DSD reproduction on FiiO devices. Tested-by: Amadej Kastelic <amadejkastelic7(a)gmail.com> Signed-off-by: Emilio Moretti <emilio.moretti(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..f5fc65aef628 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ + case 0x2972: /* FiiO devices */ case 0x2ab6: /* T+A devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ -- 2.20.1

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Dell fixed type headset platform
by Kailang 17 Dec '20

17 Dec '20

Hi Takashi, Dell has fixed type headset platform for desktop. Attach patch was add pin verb. BIOS verb table was setting for fixed type coef. Many Thanks. BR, Kailang

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[PATCH v4 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 17 Dec '20

17 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Srinivasa Rao Mandadapu (2): Partially revert ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++---------------- sound/soc/qcom/lpass-platform.c | 46 +++++++++++++++++++++++++++++++---------- 2 files changed, 37 insertions(+), 29 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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rsnd: clk-multiplier already disabled warning on Ebisu
by Geert Uytterhoeven 17 Dec '20

17 Dec '20

Hi Morimoto-san, While booting Ebisu-4D, I saw the below warnings (once). rcar_sound ec500000.sound: can't use clk 1 rsnd_adg_clk_control() failed to enable CLKB, but continues. ------------[ cut here ]------------ clk-multiplier already disabled WARNING: CPU: 1 PID: 52 at drivers/clk/clk.c:952 clk_core_disable+0x30/0x9c Modules linked in: CPU: 1 PID: 52 Comm: kworker/1:2 Not tainted 5.10.0-rc7-rcar3-initrd-00582-gadecf297acf4-dirty #183 Hardware name: Renesas Ebisu-4D board based on r8a77990 (DT) Workqueue: events deferred_probe_work_func pstate: 60000085 (nZCv daIf -PAN -UAO -TCO BTYPE=--) pc : clk_core_disable+0x30/0x9c lr : clk_core_disable+0x30/0x9c sp : ffffffc010f939f0 x29: ffffffc010f939f0 x28: 0000000000000000 x27: ffffff800949eb00 x26: ffffff80093c40f8 x25: 0000000000000001 x24: ffffffc010805fea x23: 0000000000000000 x22: ffffff80082b8810 x21: ffffff80093c4080 x20: ffffff8009368b00 x19: ffffff8009368b00 x18: 0000000000000000 x17: 000000004ef774c4 x16: 0000000000000014 x15: ffffffffffffffff x14: ffffffc010b8e8b0 x13: fffffffffff9fd47 x12: 0720072007200720 x11: fffffffffffc0000 x10: ffffffc010b8e8d8 x9 : 0720072007200720 x8 : 2079646165726c61 x7 : 0000000000000004 x6 : 00000000ffffe6ad x5 : ffffffc010f93718 x4 : 0000000000000000 x3 : 0000000000000000 x2 : 00000000ffffffff x1 : 0000000000000000 x0 : 0000000000000000 Call trace: clk_core_disable+0x30/0x9c clk_core_disable_lock+0x24/0x3c clk_disable+0x20/0x30 rsnd_adg_clk_control+0xa0/0xd4 rsnd_adg_remove+0x58/0x6c rsnd_probe+0x134/0x3cc Something else fails (-EPROBE_DEFER?), and thus rsnd_adg_remove() is called. The latter disables all clocks, including the one that failed to enable before, leading to the warning. platform_drv_probe+0x54/0xa4 really_probe+0x234/0x340 driver_probe_device+0xa0/0xb0 __device_attach_driver+0x9c/0xa8 bus_for_each_drv+0x9c/0xc4 __device_attach+0xd4/0x140 device_initial_probe+0x14/0x20 bus_probe_device+0x30/0x94 deferred_probe_work_func+0x9c/0xb0 process_one_work+0x180/0x1f0 process_scheduled_works+0x44/0x48 worker_thread+0x1ec/0x270 kthread+0xdc/0xec ret_from_fork+0x10/0x30 ---[ end trace 8d8c2a1b7537ca36 ]--- ------------[ cut here ]------------ clk-multiplier already unprepared WARNING: CPU: 1 PID: 52 at drivers/clk/clk.c:810 clk_core_unprepare+0x30/0xd0 [...] ---[ end trace 8d8c2a1b7537ca37 ]--- Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[PATCH v2] ASoC: AMD Renoir - add DMI table to avoid the ACP mic probe (broken BIOS)
by Jaroslav Kysela 17 Dec '20

17 Dec '20

Users reported that some Lenovo AMD platforms do not have ACP microphone, but the BIOS advertises it via ACPI. This patch create a simple DMI table, where those machines with the broken BIOS can be added. The DMI description for Lenovo IdeaPad 5 and IdeaPad Flex 5 devices are added there. Also describe the dmic_acpi_check kernel module parameter in a more understandable way. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index b943e59fc302..60a3e2f442db 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -6,6 +6,7 @@ #include <linux/pci.h> #include <linux/acpi.h> +#include <linux/dmi.h> #include <linux/module.h> #include <linux/io.h> #include <linux/delay.h> @@ -20,14 +21,13 @@ module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); /** - * dmic_acpi_check = -1 - Checks ACPI method to know DMIC hardware status runtime - * = 0 - Skips the DMIC device creation and returns probe failure - * = 1 - Assumes that platform has DMIC support and skips ACPI - * method check + * dmic_acpi_check = -1 - Use ACPI/DMI method to detect the DMIC hardware presence at runtime + * = 0 - Skip the DMIC device creation and return probe failure + * = 1 - Force DMIC support */ static int dmic_acpi_check = ACP_DMIC_AUTO; module_param(dmic_acpi_check, bint, 0644); -MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime"); +MODULE_PARM_DESC(dmic_acpi_check, "Digital microphone presence (-1=auto, 0=none, 1=force)"); struct acp_dev_data { void __iomem *acp_base; @@ -163,6 +163,17 @@ static int rn_acp_deinit(void __iomem *acp_base) return 0; } +static const struct dmi_system_id rn_acp_quirk_table[] = { + { + /* Lenovo IdeaPad Flex 5 14ARE05, IdeaPad 5 15ARE05 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"), + } + }, + {} +}; + static int snd_rn_acp_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -172,6 +183,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, acpi_handle handle; acpi_integer dmic_status; #endif + const struct dmi_system_id *dmi_id; unsigned int irqflags; int ret, index; u32 addr; @@ -232,6 +244,12 @@ static int snd_rn_acp_probe(struct pci_dev *pci, goto de_init; } #endif + dmi_id = dmi_first_match(rn_acp_quirk_table); + if (dmi_id && !dmi_id->driver_data) { + dev_info(&pci->dev, "ACPI settings override using DMI (ACP mic is not present)"); + ret = -ENODEV; + goto de_init; + } } adata->res = devm_kzalloc(&pci->dev, -- 2.26.2

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chtrt5645: Enable Internal MIC of ECS EF20EA
by GitHub pull_request - edited 17 Dec '20

17 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * ~~Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver.~~ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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chtrt5645: Enable Internal MIC of ECS EF20EA
by GitHub pull_request - edited 17 Dec '20

17 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: SOF: Fix spelling mistake in Kconfig "ond" -> "and"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..3e8b6c035ce3 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_SOF_DEBUG_XRUN_STOP bool "SOF stop on XRUN" help This option forces PCMs to stop on any XRUN event. This is useful to - preserve any trace data ond pipeline status prior to the XRUN. + preserve any trace data and pipeline status prior to the XRUN. Say Y if you are debugging SOF FW pipeline XRUNs. If unsure select "N". -- 2.29.2

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[PATCH] ASoC: codecs: fix spelling mistake in Kconfig "comunicate" -> "communicate"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.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 ba4eb54aafcb..9bf6bfdaf11e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -457,7 +457,7 @@ config SND_SOC_ADAU7118_HW help Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter. In this mode, the device works in standalone mode which - means that there is no bus to comunicate with it. Stereo mode is not + means that there is no bus to communicate with it. Stereo mode is not supported in this mode. To compile this driver as a module, choose M here: the module -- 2.29.2

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[PATCH] ASoC: atmel: fix spelling mistake in Kconfig "programable" -> "programmable"
by Colin King 16 Dec '20

16 Dec '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of spelling mistakes in the Kconfig help text. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 142373ec411a..9fe9471f4514 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -143,7 +143,7 @@ config SND_MCHP_SOC_SPDIFTX - sama7g5 This S/PDIF TX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. config SND_MCHP_SOC_SPDIFRX tristate "Microchip ASoC driver for boards using S/PDIF RX" @@ -157,5 +157,5 @@ config SND_MCHP_SOC_SPDIFRX - sama7g5 This S/PDIF RX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. endif -- 2.29.2

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[PATCH 0/2] ASoC: ti: Maintainer mail address change
by Peter Ujfalusi 16 Dec '20

16 Dec '20

Hi, My TI address is going to bounce after Friday (18.12.2020), switch my email address to my private one for now. Regards, Peter --- Peter Ujfalusi (2): MAINTAINERS: Update email address for TI ASoC and twl4030 codec drivers ASoC: dt-bindings: ti,j721e: Update maintainer and author information .../devicetree/bindings/sound/ti,j721e-cpb-audio.yaml | 4 +++- .../devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 4 +++- MAINTAINERS | 6 +++--- 3 files changed, 9 insertions(+), 5 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH] ASoC: dapm: remove widget from dirty list on free
by Thomas Hebb 16 Dec '20

16 Dec '20

A widget's "dirty" list_head, much like its "list" list_head, eventually chains back to a list_head on the snd_soc_card itself. This means that the list can stick around even after the widget (or all widgets) have been freed. Currently, however, widgets that are in the dirty list when freed remain there, corrupting the entire list and leading to memory errors and undefined behavior when the list is next accessed or modified. I encountered this issue when a component failed to probe relatively late in snd_soc_bind_card(), causing it to bail out and call soc_cleanup_card_resources(), which eventually called snd_soc_dapm_free() with widgets that were still dirty from when they'd been added. Fixes: db432b414e20 ("ASoC: Do DAPM power checks only for widgets changed since last run") Cc: stable(a)vger.kernel.org Signed-off-by: Thomas Hebb <tommyhebb(a)gmail.com> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f87b449f950..148c095df27b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2486,6 +2486,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) enum snd_soc_dapm_direction dir; list_del(&w->list); + list_del(&w->dirty); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both -- 2.29.2

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[PATCH] ALSA: hda/realtek: Remove dummy lineout on Acer TravelMate P648/P658
by Chris Chiu 16 Dec '20

16 Dec '20

Acer TravelMate laptops P648/P658 series with codec ALC282 only have one physical jack for headset but there's a confusing lineout pin on NID 0x1b reported. Audio applications hence misunderstand that there are a speaker and a lineout, and take the lineout as the default audio output. Add a new quirk to remove the useless lineout and enable the pin 0x18 for jack sensing and headset microphone. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..e7edcdbc6ca3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6366,6 +6366,7 @@ enum { ALC287_FIXUP_HP_GPIO_LED, ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, + ALC282_FIXUP_ACER_DISABLE_LINEOUT, }; static const struct hda_fixup alc269_fixups[] = { @@ -7789,6 +7790,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ACER_DISABLE_LINEOUT] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x411111f0 }, + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8560,6 +8571,22 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60140}, {0x19, 0x04a11030}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0282, 0x1025, "Acer", ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC282_STANDARD_PINS, + {0x12, 0x90a609c0}, + {0x18, 0x03a11830}, + {0x19, 0x04a19831}, + {0x1a, 0x0481303f}, + {0x1b, 0x04211020}, + {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0282, 0x1025, "Acer", ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC282_STANDARD_PINS, + {0x12, 0x90a60940}, + {0x18, 0x03a11830}, + {0x19, 0x04a19831}, + {0x1a, 0x0481303f}, + {0x1b, 0x04211020}, + {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, -- 2.20.1

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[PATCH] ASoC: Intel: Skylake: skl-topology: Fix OOPs ib skl_tplg_complete
by Ricardo Ribalda 15 Dec '20

15 Dec '20

If dobj->control is not initialized we end up in an OOPs during skl_tplg_complete: [ 26.553358] BUG: kernel NULL pointer dereference, address: 0000000000000078 [ 26.561151] #PF: supervisor read access in kernel mode [ 26.566897] #PF: error_code(0x0000) - not-present page [ 26.572642] PGD 0 P4D 0 [ 26.575479] Oops: 0000 [#1] PREEMPT SMP PTI [ 26.580158] CPU: 2 PID: 2082 Comm: udevd Tainted: G C 5.4.81 #4 [ 26.588232] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.106.0 12/03/2019 [ 26.597082] RIP: 0010:skl_tplg_complete+0x70/0x144 [snd_soc_skl] Fixes: 2d744ecf2b98 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHL") Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- sound/soc/intel/skylake/skl-topology.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 40bee10b0c65..0955cbb4e918 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3619,19 +3619,20 @@ static void skl_tplg_complete(struct snd_soc_component *component) list_for_each_entry(dobj, &component->dobj_list, list) { struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct soc_enum *se; + char **texts; char chan_text[4]; - if (dobj->type != SND_SOC_DOBJ_ENUM || - dobj->control.kcontrol->put != - skl_tplg_multi_config_set_dmic) + if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || + kcontrol->put != skl_tplg_multi_config_set_dmic) continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; + struct snd_ctl_elem_value val = {}; if (strstr(texts[i], chan_text)) { val.value.enumerated.item[0] = i; -- 2.29.2.576.ga3fc446d84-goog

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Enable ECS EF20EA's internal MIC and HDMI audio
by GitHub pull_request - edited 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was edited from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Enable ECS EF20EA's internel MIC and HDMI audio
by GitHub pull_request - opened 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #68 was opened from starnight: * Add a card long name "Standard-EF20EA-1.0" match chtrt5645 enabling the internal DMIC. * Create HdmiLpeAudio folder and copy & modify HDMI.conf from HDA-Intel for the devices using HdmiLpeAudio as the HDMI audio driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/68.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire - add support for new main capture switch and main capture volume control
by GitHub pull_request - opened 15 Dec '20

15 Dec '20

alsa-project/alsa-ucm-conf pull request #67 was opened from GoPerry: This patch was added for its new main capture switch and main capture volume control which combine the ADC 07 / ADC 27 Capture and Volume switch ,the new mute switch and volume control will be handled by codec rt715`s driver mute/volume kcontrol instead of PGA5. BugLink: https://github.com/thesofproject/linux/issues/2544 Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/67 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/67.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Intel: fix error code cnl_set_dsp_D0()
by Dan Carpenter 14 Dec '20

14 Dec '20

Return -ETIMEDOUT if the dsp boot times out instead of returning success. Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Not tested. sound/soc/intel/skylake/cnl-sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index fcd8dff27ae8..1275c149acc0 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -224,6 +224,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) "dsp boot timeout, status=%#x error=%#x\n", sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); + ret = -ETIMEDOUT; goto err; } } else { -- 2.29.2

3 2

[RFC PATCH] ASoC: pcm_dmaengine: Add support for BE DAIs
by Codrin Ciubotariu 14 Dec '20

14 Dec '20

Considering the bellow scenarios: /- Gen DMA engine -\ 48KHz ******** ******* 44.1KHz ******** ******* ********* PCM0 <-FE DAI-> * FE * <-> * DSP * <-BE DAI-> * * * * * * * Ring * * * * Ring * <-> * CPU * <-> * Codec * * buff * ******* * buff * * DAI * * DAI * 44.1KHz ******** * * * * * * PCM1 <--------------------DAI-----------------> * * * * * * ******** ******* ********* For PCM0, we have two DAI links. The first DAI link is a FE, with a DSP as a CPU DAI and a platform driver .The second DAI link is a BE DAI link, with separate CPU, codec and platform drivers. We can also notice that there are two Ring buffers: the first one used by the DSP to communicate with the user-space and the second one used to move data between FE (DSP) and BE (CPU). PCM1 is a normal DAI link, with a CPU, codec and platform driver. It is exactly the previous BE DAI link from PCM0, so the samples from user-space are copied directly into the second Ring buffer. In this scenario, the BE DAI link driver should be the same, since it is decided at runtime whether the DAI is a used as a BE or not. The generic platform driver needs to be aware of this thing. For the BE case (PCM0), some callbacks of the platform driver are not called, hence the preallocated buffer is not available. Also, the PCM runtime strcture must not be touched, since it should be only used by the FE platform driver. With these changes, the generic dmaengine can also be used as a BE platform driver. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Hello, This patch is more or less incomplete for the described scenario. This is because DMAengine's pcm->config is ignored for the BE DAI link, so runtime->hw is not updated. Also, since pcm_construct/destruct are not called, the DMA channels are allocated only if DT is used. Underrun/overrun support would also be a nice to have for the transfers involving the buffer allocated for the BE. One way to hold trach of these would be to use a substream_be->runtime different than the one used for the FE. Please share your thoughts. sound/core/pcm_dmaengine.c | 18 ++++++++++-- sound/soc/soc-generic-dmaengine-pcm.c | 40 +++++++++++++++++++++++---- 2 files changed, 50 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d059ff2b2e4..5e96bc27628d 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -137,7 +137,9 @@ static void dmaengine_pcm_dma_complete(void *arg) if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) prtd->pos = 0; - snd_pcm_period_elapsed(substream); + /* do no update period for an internal PCM */ + if (!substream->pcm->internal) + snd_pcm_period_elapsed(substream); } static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) @@ -147,6 +149,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) struct dma_async_tx_descriptor *desc; enum dma_transfer_direction direction; unsigned long flags = DMA_CTRL_ACK; + dma_addr_t addr; direction = snd_pcm_substream_to_dma_direction(substream); @@ -154,11 +157,15 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) flags |= DMA_PREP_INTERRUPT; prtd->pos = 0; + if (substream->pcm->internal) + addr = substream->dma_buffer.addr; + else + addr = substream->runtime->dma_addr; + desc = dmaengine_prep_dma_cyclic(chan, - substream->runtime->dma_addr, + addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), direction, flags); - if (!desc) return -ENOMEM; @@ -315,6 +322,11 @@ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, substream->runtime->private_data = prtd; + if (substream->pcm->internal) { + substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_IRAM; + substream->dma_buffer.dev.dev = chan->device->dev; + } + return 0; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 9ef80a48707e..f403849cd1aa 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -97,6 +97,31 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, if (ret) return ret; } + if (!substream->pcm->internal) { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + } + + /* allocate a buffer for BE DAI; for now, the buffer will have the same + * size as the buffer used by the FE + */ + if (snd_dma_alloc_pages(substream->dma_buffer.dev.type, + substream->dma_buffer.dev.dev, + params_buffer_bytes(params), + &substream->dma_buffer) < 0) + return -ENOMEM; + + return 0; +} + +static int dmaengine_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (!substream->pcm->internal) + return 0; + + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; return 0; } @@ -157,9 +182,12 @@ static int dmaengine_pcm_open(struct snd_soc_component *component, struct dma_chan *chan = pcm->chan[substream->stream]; int ret; - ret = dmaengine_pcm_set_runtime_hwparams(component, substream); - if (ret) - return ret; + /* do not touch runtime if this is an internal PCM */ + if (!substream->pcm->internal) { + ret = dmaengine_pcm_set_runtime_hwparams(component, substream); + if (ret) + return ret; + } return snd_dmaengine_pcm_open(substream, chan); } @@ -309,7 +337,7 @@ static int dmaengine_copy_user(struct snd_soc_component *component, channel * (runtime->dma_bytes / runtime->channels); int ret; - if (is_playback) + if (!substream->pcm->internal && is_playback) if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; @@ -319,7 +347,7 @@ static int dmaengine_copy_user(struct snd_soc_component *component, return ret; } - if (!is_playback) + if (!substream->pcm->internal && !is_playback) if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; @@ -332,6 +360,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = { .open = dmaengine_pcm_open, .close = dmaengine_pcm_close, .hw_params = dmaengine_pcm_hw_params, + .hw_free = dmaengine_pcm_hw_free, .trigger = dmaengine_pcm_trigger, .pointer = dmaengine_pcm_pointer, .pcm_construct = dmaengine_pcm_new, @@ -344,6 +373,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component_process = { .close = dmaengine_pcm_close, .hw_params = dmaengine_pcm_hw_params, .trigger = dmaengine_pcm_trigger, + .hw_free = dmaengine_pcm_hw_free, .pointer = dmaengine_pcm_pointer, .copy_user = dmaengine_copy_user, .pcm_construct = dmaengine_pcm_new, -- 2.27.0

3 5

[GIT PULL] ASoC updates for v5.11
by Mark Brown 14 Dec '20

14 Dec '20

The following changes since commit aa4cb898b80a28a610e26d1513e6dd42d995c225: ASoC: rt5682: change SAR voltage threshold (2020-11-26 13:06:18 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.11 for you to fetch changes up to 460aa020f56c974a3e7e5b5378b2355fec6a2c11: Merge remote-tracking branch 'asoc/for-5.11' into asoc-next (2020-12-11 17:48:04 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.11 There's a lot of changes here but mostly cleanups and driver specific things, the most user visible change is the support for boot time selection of Intel DSP firmware which will make it easier for people to move over to the preferred modern implementations in distros and other large scale deployments. This also includes a merge of the new auxillary bus which was done in anticipation of use by the Intel DSP drivers which didn't quite make it. - Lots more cleanups and simplifications from Morimoto-san. - Support for some basic DPCM systems in the audio graph card from Sameer Pujar. - Remove some old pre-DT Freescale drivers for platforms that are now DT only. - Move selection of which Intel DSP implementation to use to boot time rather than requiring it to be selected at build time. - Support for Allwinner H6 I2S, Analog Devices ADAU1372, Intel Alderlake-S, GMediatek MT8192, NXP i.MX HDMI and XCVR, Realtek RT715, Qualcomm SM8250 and simple GPIO based muxes. ---------------------------------------------------------------- Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic Alexandre Belloni (4): ASoC: adau1372: Add bindings documentation ASoC: adau1372: add missing dependencies ASoC: add simple-audio-mux binding ASoC: add simple-mux Alexandru Ardelean (2): ASoC: adau1977: remove platform data and move micbias bindings include ASoC: adau1977: convert text binding to yaml format Amadeusz Sławiński (8): ASoC: topology: Remove unused functions from topology API ASoC: topology: Remove multistep topology loading ASoC: topology: Unify all device references ASoC: topology: Change allocations to resource managed ASoC: topology: Remove empty functions ASoC: topology: Simplify remove_widget function ASoC: topology: Add missing size check ASoC: topology: Fix wrong size check Arnd Bergmann (4): ASoC: cros_ec_codec: fix uninitialized memory read ASoC: atmel: mchp-spdifrx needs COMMON_CLK ASoC: fsl_aud2htx: mark PM functions as __maybe_unused ASoC: qcom: fix QDSP6 dependencies, attempt #3 Bogdan Togorean (1): ASoc: adi: Kconfig: Remove depends on for ADI reference designs Brent Lu (3): ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682 ASoC: intel: sof_rt5682: Add quirk for Dooly ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682 Cezary Rojewski (4): ASoC: pcm: DRAIN support reactivation ASoC: Intel: catpt: Optimize applying user settings ASoC: Intel: catpt: Streamline power routines across LPT and WPT ASoC: Intel: catpt: Cleanup after power routines streamlining Cheng-Yi Chiang (3): ASoC: google: dt-bindings: Add sc7180-trogdor machine bindings ASoC: qcom: sc7180: Add missing PM ops ASoC: hdmi-codec: Get ELD in before reporting plugged event Chris Chiu (1): ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140 Christophe Branchereau (2): ASoC: codecs/jz4770: Reset interrupt flags in bias PREPARE ASoC: codecs/jz4770: Adjust timeouts for cap-coupled outputs Chuhong Yuan (2): ASoC: jz4740-i2s: add missed checks for clk_get() ASoC: amd: change clk_get() to devm_clk_get() and add missed checks Clément Péron (6): ASoC: sun4i-i2s: Fix lrck_period computation for I2S justified mode ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition ASoC: sun4i-i2s: Document H3 with missing RX channel possibility Codrin Ciubotariu (2): ASoC: pcm5102a: Make codec selectable ASoC: atmel-i2s: do not warn if muxclk is missing Colin Ian King (1): ASoC: qcom: fix unsigned int bitwidth compared to less than zero Curtis Malainey (1): ASoC: SOF: IPC: fix implicit type overflow Dan Carpenter (7): ASoC: qcom: sc7180: Fix some indenting in sc7180_lpass_alloc_dma_channel() ASoC: qcom: common: Fix refcounting in qcom_snd_parse_of() ASoC: qcom: sc7180: initialize the "no_headphone" variable ASoC: codecs: lpass-va-macro: remove some dead code ASoC: max98390: Fix error codes in max98390_dsm_init() ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control() ASoC: mediatek: mt8183: delete some unreachable code Dave Ertman (1): Add auxiliary bus support David Lin (2): ASoC: nau8315: add codec driver ASoC: nau8315: revise the power event of EN_PIN dapm widget for symmetry Derek Fang (1): ASoC: rt1015: check the return value of regmap_read during i2c probe Dharageswari R (1): ASoC: Intel: Boards: tgl_max98373: add dpcm_capture flag for speaker_smart_amp Dmitry Baryshkov (1): ASoC: qcom: sm8250: fix HDMI audio playback Fabio Estevam (8): ASoC: wm1133-ev1: Remove unused driver ASoC: mx27vis-aic32x4: Remove unused driver ASoC: phycore-ac97: Remove unused driver ASoC: imx-mc13783: Remove unused driver ASoC: fsl: eukrea: Remove the SND_SOC_IMX_SSI selection ASoC: imx-ssi: Remove unused driver ASoC: imx-audmux: Remove unused .id_table ASoC: fsl_audmix: Remove unneeded data field Fred Oh (5): ASoC: SOF: ops: add parse_platform_ext_manifest() op ASoC: SOF: Intel: hda: define parse_platform_ext_manifest op ASoC: SOF: ext_manifest: parse cavs extra config data elem ASoC: SOF: ops: modify the signature of stall op ASoC: SOF: Intel: hda: add sof_icl_ops for ICL platforms Geert Uytterhoeven (3): ASoC: fsl: SND_SOC_FSL_AUD2HTX should depend on ARCH_MXC ASoC: intel: SND_SOC_INTEL_KEEMBAY should depend on ARCH_KEEMBAY ASoC: Fix 7/8 spaces indentation in Kconfig Greg Kroah-Hartman (3): driver core: auxiliary bus: move slab.h from include file driver core: auxiliary bus: make remove function return void driver core: auxiliary bus: minor coding style tweaks Gustavo A. R. Silva (1): ASoC: codecs: Fix fall-through warnings for Clang Hans de Goede (2): ASoC: Intel: cht_bsw_nau8824: Drop compress-cpu-dai bits ASoC: Intel: cht_bsw_nau8824: Change SSP2-Codec DAI id to 0 Jack Yu (3): ASoC: rt1015: modification for calibration to get better performance ASoC/SoundWire: rt715-sdca: First version of rt715 sdw sdca codec driver ASoC: rt715: remove unused parameter Jaroslav Kysela (2): ASoC: AMD Raven/Renoir - fix the PCI probe (PCI revision) ASoC: AMD Renoir - add DMI table to avoid the ACP mic probe (broken BIOS) Jaska Uimonen (1): ASoC: SOF: control: override volume info callback Jernej Skrabec (2): ASoC: sun4i-i2s: Add support for H6 I2S ASoC: sun4i-i2s: Add H6 compatible Jerome Brunet (1): ASoC: meson: fix COMPILE_TEST error Jiaxin Yu (13): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver ASoC: mediatek: mt8192: support i2s in platform driver ASoC: mediatek: mt8192: support adda in platform driver ASoC: mediatek: mt8192: support pcm in platform driver ASoC: mediatek: mt8192: support tdm in platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document ASoC: mediatek: mt8192: skip first time data at the beginning of DMIC recording ASoC: mt8192: revert "add audio afe document" ASoC: mediatek: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18" ASoC: mt6359: remove unused property for mt6359 Jon Hunter (1): ASoC: tegra: Don't warn on probe deferral Julia Lawall (6): ASoC: wm8350: use semicolons rather than commas to separate statements ASoC: Intel: bytcr_rt5651: use semicolons rather than commas to separate statements ASoC: SOF: Intel: hda: use semicolons rather than commas to separate statements ASoC: samsung: snow: use semicolons rather than commas to separate statements ASoC: madera: use semicolons rather than commas to separate statements ASoC: dapm: use semicolons rather than commas to separate statements Kai Vehmanen (6): ASoC: SOF: loader: do not warn about unknown firmware headers ASoC: SOF: Intel: initial support for Alderlake-S ASoC: SOF: relax PCM period and buffer size constraints ASoC: SOF: Intel: add hw specific PCM constraints ASoC: Intel: common: add ACPI matching tables for Alder Lake ASoC: SOF: Intel: add SoundWire support for ADL-S Kaixu Xia (1): ASoC: wcd9335: Remove unnecessary conversion to bool Karol Trzcinski (5): ASoC: SOF: ext_manifest: Parse firmware config dictionary ASoC: SOF: Improve code alignment in header.h ASoC: SOF: Change section comment for SOF_IPC_TEST_ ASoC: SOF: Add `memory_info` file to debugfs ASoC: SOF: trace: Add runtime trace filtering mechanism Kirill Marinushkin (4): ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS ASoC: pcm512x: Rearrange operations in `hw_params()` ASoC: pcm512x: Move format check into `set_fmt()` ASoC: pcm512x: Add support for more data formats Krzysztof Kozlowski (43): ASoC: sunxi: do not select COMMON_CLK to fix builds ASoC: sh: depend on COMMON_CLK to fix compile tests ASoC: stm: depend on COMMON_CLK to fix compile tests ASoC: ti: depend on COMMON_CLK to fix compile tests ASoC: ak5558: mark OF related data as maybe unused ASoC: bd28623: mark OF related data as maybe unused ASoC: gtm601: mark OF related data as maybe unused ASoC: inno_rk3036: mark OF related data as maybe unused ASoC: rk3328: mark OF related data as maybe unused ASoC: tas571x: mark OF related data as maybe unused ASoC: kirkwood: armada-370-db: mark OF related data as maybe unused ASoC: meson: t9015: mark OF related data as maybe unused ASoC: qcom: mark OF related data as maybe unused ASoC: samsung: smdk_wm8994: mark OF related data as maybe unused ASoC: rockchip: mark OF related data as maybe unused ASoC: ti: davinci: mark OF related data as maybe unused ASoC: uniphier: mark OF related data as maybe unused ASoC: ak4118: skip of_device_id table when !CONFIG_OF ASoC: alc5623: skip of_device_id table when !CONFIG_OF ASoC: alc5632: skip of_device_id table when !CONFIG_OF ASoC: da7218: skip of_device_id table when !CONFIG_OF ASoC: da7219: skip of_device_id table when !CONFIG_OF ASoC: da9055: skip of_device_id table when !CONFIG_OF ASoC: es8316: skip of_device_id table when !CONFIG_OF ASoC: max98090: skip of_device_id table when !CONFIG_OF ASoC: max98095: skip of_device_id table when !CONFIG_OF ASoC: max98371: skip of_device_id table when !CONFIG_OF ASoC: max9867: skip of_device_id table when !CONFIG_OF ASoC: max98925: skip of_device_id table when !CONFIG_OF ASoC: max98926: skip of_device_id table when !CONFIG_OF ASoC: pcm1789: skip of_device_id table when !CONFIG_OF ASoC: pcm179x: skip of_device_id table when !CONFIG_OF ASoC: rt5660: skip of_device_id table when !CONFIG_OF ASoC: tas2562: skip of_device_id table when !CONFIG_OF ASoC: tlv320: skip of_device_id table when !CONFIG_OF ASoC: ts3a227e: skip of_device_id table when !CONFIG_OF ASoC: es7134: mark OF related data as maybe unused ASoC: es7241: mark OF related data as maybe unused ASoC: samsung: i2s: mark OF related data as maybe unused ASoC: max98371: drop driver pm=NULL assignment ASoC: max98925: drop driver pm=NULL assignment ASoC: max98926: drop driver pm=NULL assignment ASoC: samsung: smdk_wm8994: remove redundant of_match_ptr() Kuninori Morimoto (34): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() ASoC: audio-graph-card: switch to yaml base Documentation ASoC: renesas, rsnd: switch to yaml base Documentation ASoC: renesas,rsnd: Add r8a77961 support ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() ASoC: soc-compress: move soc_compr_free() next to soc_compr_open() ASoC: soc-dai: add mark for snd_soc_dai_compr_startup/shutdown() ASoC: soc-component: add mark for snd_soc_component_compr_open/free() ASoC: soc-component: add mark for snd_soc_link_compr_startup/shutdown() ASoC: soc-compress: add soc_compr_clean() and call it from soc_compr_open/free() ASoC: soc-core: add soc_playback_digital_mute() ASoC: soc-core: add soc_dapm_suspend_resume() ASoC: soc-core: tidyup jack.h ASoC: add soc-jack.h ASoC: soc-pcm: remove dpcm_do_trigger() ASoC: soc-pcm: care trigger rollback Kyle Russell (2): ASoC: mmp-sspa: clear transmit phase bit for non-stereo formats ASoC: mmp-sspa: set phase two word length register Lars-Peter Clausen (2): ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment ASoC: Add ADAU1372 audio CODEC support Lumi Lee (1): ASoC: mediatek: btcvsd fix tx stream assign Marcus Cooper (2): ASoC: sun4i-i2s: Set sign extend sample ASoC: sun4i-i2s: Add 20 and 24 bit support Mark Brown (48): Merge existing fixes from asoc/for-5.10 Merge series "dt-bindings: stm32: convert audio dfsdm to json-schema" from Olivier Moysan <olivier.moysan(a)st.com>: Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel(a)sholland.org>: Merge series "use semicolons rather than commas to separate statements" from Julia Lawall <Julia.Lawall(a)inria.fr>: Merge series "ASoC: qcom: add support for QRB5165 RB5 machine" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "DAI driver for new XCVR IP" from "Viorel Suman (OSS)" <viorel.suman(a)oss.nxp.com> Viorel Suman <viorel.suman(a)nxp.com>: Merge series "ASoC: SOF: cleanups for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "Add documentation and machine driver for SC7180 sound card" from Cheng-Yi Chiang <cychiang(a)chromium.org>: Merge series "Add rt1015 support to CML boards" from Brent Lu <brent.lu(a)intel.com>: Merge series "ASoC: Mediatek: Add support for MT8192 SoC" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: topology: Change to resource managed memory" from Amadeusz Sławiński<amadeuszx.slawinski(a)linux.intel.com>: Merge series "ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully" from Peter Ujfalusi <peter.ujfalusi(a)ti.com>: Merge series "ASoC: soc-compress: tidyup STREAM vs COMPRESS" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Fix vaud18 power leakage of mt6359" from Shane Chien <shane.chien(a)mediatek.com> "Shane.Chien" <shane.chien(a)mediatek.com>: Merge series "Modify documentation and machine driver for SC7180 sound card" from Ajye Huang <ajye.huang(a)gmail.com>: Merge series "Audio Graph Updates" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "Fix PM disable depth imbalance on error" from Zhang Qilong <zhangqilong3(a)huawei.com>: Merge series "ASoC: Fix error handling in wm899x" from Zhang Qilong <zhangqilong3(a)huawei.com>: Merge series "ASoC: soundwire: SDCA device quirk and cleanups" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Support SC7180 sound card variation without headset" from xuyuqing <xuyuqing(a)huaqin.corp-partner.google.com>: Merge series "ASoC: SOF: Kconfig corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: use inclusive language for bclk/fsync/topology" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: pcm512x: Patch series to set fmt from `set_fmt()`" from Kirill Marinushkin <kmarinushkin(a)birdec.com>: Merge series "ASoC: Intel: catpt: Offload fixes and code optimization" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge series "ASoC: graph card schema rework" from Rob Herring <robh(a)kernel.org>: Merge series "ASoC: soc-component: add snd_soc_component_xxx()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: codecs: add support for LPASS Codec macros" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: Intel/SOF: extend run-time driver selection to ACPI devices" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18"" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: fix !OF compile test warnings" from Krzysztof Kozlowski <krzk(a)kernel.org>: Merge series "ASoC: merge soc_compr_open() rollback and soc_compr_free()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: amd: Minor fixes for error handling" from Takashi Iwai <tiwai(a)suse.de>: Merge series "ASoC: tidyup for jack.h" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: SOF: Intel: fix ICL boot sequence" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge branch 'for-5.10' of https://git.kernel.org/.../broonie/sound into asoc-5.11 Merge tag 'soundwire-for-asoc-5.11' of git://git.kernel.org/.../vkoul/soundwire into asoc-5.11 Merge branch '20201104_yung_chuan_liao_regmap_soundwire_asoc_add_soundwire_sdca_support' (early part) into asoc-5.11 Merge series "ASoC: mediatek: mt8192: support new machine rt1015p_rt5682" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge tag 'auxbus-5.11-rc1' of git://git.kernel.org/.../gregkh/driver-core into asoc-5.11 Merge series "ASoC: Intel: cht_bsw_nau8824: 2 fixes for usage with sof-audio-acpi" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: soc-pcm: trigger cleanup" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rt1015p: delay 300ms for waiting calibration" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: SOF: Intel: fix to dsp state dump trace levels" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge remote-tracking branch 'asoc/for-5.9' into asoc-linus Merge remote-tracking branch 'asoc/for-5.10' into asoc-linus Merge remote-tracking branch 'asoc/for-5.11' into asoc-next Michał Mirosław (1): ASoC: tegra20-spdif: remove "default m" Oder Chiou (2): ASoC: rt5682: Add a new property for the DMIC clock driving ASoC: rt5682: Add a new property to the device tree. Olivier Moysan (3): dt-bindings: stm32: dfsdm: update audio properties ASoC: dt-bindings: stm32: dfsdm: remove stm32-adfsdm.txt binding ASoC: dt-bindings: stm32: convert sai to json-schema Paul Cercueil (3): ASoC: codecs/jz47xx: Use regmap_{set,clear}_bits ASoC: codecs/jz4770: Don't change cap-couple setting in HP PMU/PMD ASoC: codecs/jz4770: Add DAPM widget to set HP out to cap-less mode Peter Ujfalusi (4): ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling ASoC: ti: davinci-mcasp: Handle missing required DT properties Pierre-Louis Bossart (31): ASoC: SOF: control: remove const in sizeof() ASoC: SOF: topology: remove const in sizeof() ASoC: SOF: sof-audio: remove goto used for force-nocodec support ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device ASoC: codecs: rt*.c: remove useless pointer cast ASoC: rt711: remove warnings ASoC: codecs: max98373-sdw: align regmap use with other codecs ASoC: SOF: Intel: fix Kconfig dependency for SND_INTEL_DSP_CONFIG ASoC: topology: use inclusive language for bclk and fsync ASoC: SOF: use inclusive language for bclk and fsync ASoC: Intel: atom: use inclusive language for SSP bclk/fsync ASoC: Intel: keembay: use inclusive language for bclk and fsync ASoC: Intel: broadwell: add missing pm_ops ASoC: Intel: bdw-rt5677: add missing pm_ops ALSA: hda: intel-dsp-config: add helper for ACPI DSP driver selection ASoC: soc-acpi: add helper to identify parent driver. ASoC: Intel: boards: byt/cht: set card and driver name at run time ASoC: Intel: byt/cht: set pm ops dynamically ASoC: SOF: acpi: add dynamic selection of DSP driver ASoC: Intel: Atom: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and Atom/SST drivers ALSA: hda: intel-dsp-config: add Broadwell ACPI DSP driver selection ASoC: Intel: broadwell: set card and driver name dynamically ASoC: Intel: catpt: add dynamic selection of DSP driver ALSA: hda: intel-dsp-config: ignore dsp_driver parameter for PCI legacy devices ASoC: SOF: Intel: allow for coexistence between SOF and catpt drivers soundwire: SDCA: detect sdca_cascade interrupt soundwire: SDCA: add helper macro to access controls regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ ASoC: SOF: control: fix cppcheck warning in snd_sof_volume_info() ASoC: SOF: imx: update kernel-doc description Piotr Maziarz (1): ASoC: Intel: catpt: select WANT_DEV_COREDUMP Randy Dunlap (3): ASoC: SOF: imx: fix Kconfig punctuation ASoC: SOF: Kconfig: fix Kconfig punctuation and wording ASoC: SOF: Intel: fix Kconfig punctuation and wording Ranjani Sridharan (6): ASoC: pcm: call snd_soc_dapm_stream_stop() in soc_pcm_hw_clean ASoC: SOF: nocodec: modify DAI link definitions ASoC: pcm: send DAPM_STREAM_STOP event in dpcm_fe_dai_shutdown ASoC: SOF: Intel: hda: remove duplicated status dump ASoC: SOF: modify the SOF_DBG flags ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err Richard Fitzgerald (1): ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff Rob Herring (3): ASoC: dt-bindings: marvell, mmp-sspa: Use audio-graph-port schema ASoC: dt-bindings: Refine 'frame-master' and 'bitclock-master' type ASoC: dt-bindings: renesas, rsnd: Fix duplicate 'allOf' entries Sameer Pujar (10): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: audio-graph: Expose new members for asoc_simple_priv ASoC: audio-graph: Expose helpers from audio graph ASoC: audio-graph-card: Refactor schema dt-bindings: tegra: Convert HDA doc to json-schema dt-bindings: tegra: Add missing HDA properties Samuel Holland (18): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes ASoC: sun4i-i2s: Fix setting of FIFO modes Sathyanarayana Nujella (1): ASoC: Intel: Boards: tgl_max98373: update TDM slot_width Shane.Chien (3): ASoC: Fix vaud18 power leakage of mt6359 dt-bindings: mediatek: mt6359: Add new property for mt6359 ASoC: Remove mt6359_platform_driver_remove Shengjiu Wang (13): ASoC: dt-bindings: fsl_spdif: Add new compatible string for i.MX8QM ASoC: fsl_spdif: Add support for i.MX8QM platform ASoC: fsl_spdif: Add support for higher sample rates ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module ASoC: fsl_aud2htx: Add aud2htx module driver ASoC: fsl_aud2htx: Remove dev_err() usage after platform_get_irq() ASoC: fsl_sai: Correct the clock source for mclk0 ASoC: hdmi-codec: Add RX support ASoC: fsl: Fix config name of CONFIG_ARCH_MXC ASoC: fsl-asoc-card: Add support for si476x codec ASoC: bindings: fsl-asoc-card: add compatible string for si476x codec ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver ASoC: fsl: Add imx-hdmi machine driver Shuming Fan (1): ASoC: rt1015: support TDM slot configuration Srinivas Kandagatla (17): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support ASoC: qcom: qdsp6: make use of devm_of_platform_populate ASoC: qcom: sm8250: Fix array out of bounds access ASoC: qcom: dt-bindings: sm8250: update compatibles ASoC: qcom: sm8250: update compatible with new bindings ASoC: qcom: lpass-cpu: fix warning on symbol scope ASoC: q6afe-clocks: fix warning on symbol scope ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes ASoC: codecs: lpass-va-macro: add missing MODULE_DEVICE_TABLE soundwire: Fix DEBUG_LOCKS_WARN_ON for uninitialized attribute ASoC: q6afe-clocks: Add missing parent clock rate Sudip Mukherjee (1): ASoC: mediatek: mt8192: Fix build failure Takashi Iwai (3): ASoC: amd: Downgrade print level for invalid ACP mode ASoC: amd: Return -ENODEV for non-existing ACPI call ASoC: cx2072x: Fix doubly definitions of Playback and Capture streams Tom Rix (5): ASoC: bcm2835-i2s: remove unneeded semicolon ASoC: Intel: remove unneeded semicolon ASoC: TSCS454: remove unneeded semicolon ASoC: tegra: remove unneeded semicolon ASoC: TSCS42xx: remove unneeded semicolon Tzung-Bi Shih (9): ASoC: mediatek: mt8192: rename common symbols ASoC: mediatek: mt8192: extract rt1015_rt5682 specific DAI link ASoC: mediatek: mt8192: move rt1015_rt5682 specific data ASoC: dt-bindings: mt8192-mt6359: fix typo in the example ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p ASoC: mediatek: mt8192: support rt1015p_rt5682 ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: rt1015p: delay 300ms after SDB pulling high for calibration ASoC: mediatek: mt8183: add PM ops to machine drivers V Sujith Kumar Reddy (2): ASoC: qcom: lpass-sc7180: Add 32 bit format support for capture ASoC: qcom: sc7180: Register shutdown handler for lpass platform Vijendar Mukunda (1): ASoC: amd: support other audio modes for raven Viorel Suman (4): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR ASoC: fsl_xcvr: fix break condition ASoC: fsl_xcvr: fix potential resource leak Xu Wang (3): ASoC: ti: davinci-evm: Remove redundant null check before clk_disable_unprepare ASoC: samsung: i2s: Remove redundant null check before clk_disable_unprepare ASoC: pxa: pxa-ssp: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare Zhang Qilong (4): ASoC: ti: davinci-mcasp: remove always zero of davinci_mcasp_get_dt_params ASoC: wm8994: Fix PM disable depth imbalance on error ASoC: wm8998: Fix PM disable depth imbalance on error ASoC: arizona: Fix a wrong free in wm8997_probe Zou Wei (2): ASoC: mediatek: mt8192: Make some symbols static ASoC: samsung: fix platform_no_drv_owner.cocci warnings xuyuqing (3): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Modify machine driver for sound card ASoC: qcom: sc7180: fix 32 bit format for adau7002 CREDITS | 9 + Documentation/ABI/README | 10 +- Documentation/ABI/obsolete/sysfs-class-dax | 8 +- .../ABI/obsolete/sysfs-driver-hid-roccat-pyra | 3 + Documentation/ABI/obsolete/sysfs-gpio | 2 + Documentation/ABI/removed/devfs | 1 + Documentation/ABI/removed/raw1394 | 1 + Documentation/ABI/removed/sysfs-class-rfkill | 2 +- Documentation/ABI/removed/video1394 | 1 + Documentation/ABI/stable/firewire-cdev | 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sound/soc/codecs/jz4725b.c | 26 +- sound/soc/codecs/jz4740.c | 20 +- sound/soc/codecs/jz4770.c | 109 +- sound/soc/codecs/lpass-va-macro.c | 1497 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2464 +++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 17 + sound/soc/codecs/madera.c | 4 +- sound/soc/codecs/max98090.c | 2 + sound/soc/codecs/max98095.c | 2 + sound/soc/codecs/max98371.c | 3 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/max98390.c | 2 + sound/soc/codecs/max9867.c | 2 + sound/soc/codecs/max98925.c | 3 +- sound/soc/codecs/max98926.c | 3 +- sound/soc/codecs/mt6359.c | 152 +- sound/soc/codecs/mt6359.h | 15 +- sound/soc/codecs/nau8315.c | 166 ++ sound/soc/codecs/pcm1789-i2c.c | 2 + sound/soc/codecs/pcm179x-i2c.c | 2 + sound/soc/codecs/pcm512x.c | 134 +- sound/soc/codecs/rk3328_codec.c | 2 +- sound/soc/codecs/rt1015.c | 211 +- sound/soc/codecs/rt1015.h | 42 + sound/soc/codecs/rt1015p.c | 69 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5660.c | 2 + sound/soc/codecs/rt5682-i2c.c | 5 + sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 3 + sound/soc/codecs/rt5682.h | 14 + sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdw.c | 15 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 278 ++ sound/soc/codecs/rt715-sdca-sdw.h | 170 ++ sound/soc/codecs/rt715-sdca.c | 936 ++++++ sound/soc/codecs/rt715-sdca.h | 124 + sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/rt715.h | 1 - sound/soc/codecs/simple-mux.c | 124 + sound/soc/codecs/tas2562.c | 2 + sound/soc/codecs/tas571x.c | 4 +- sound/soc/codecs/tlv320adcx140.c | 2 + sound/soc/codecs/tlv320aic23-i2c.c | 2 + sound/soc/codecs/ts3a227e.c | 2 + sound/soc/codecs/tscs42xx.c | 6 +- sound/soc/codecs/tscs454.c | 8 +- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm8350.c | 3 +- sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm8994.c | 6 +- sound/soc/codecs/wm8997.c | 2 + sound/soc/codecs/wm8998.c | 4 +- sound/soc/codecs/wm_adsp.c | 5 +- sound/soc/fsl/Kconfig | 77 +- sound/soc/fsl/Makefile | 16 +- sound/soc/fsl/fsl-asoc-card.c | 13 + sound/soc/fsl/fsl_aud2htx.c | 308 ++ sound/soc/fsl/fsl_aud2htx.h | 67 + sound/soc/fsl/fsl_audmix.c | 13 +- sound/soc/fsl/fsl_sai.c | 20 +- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/fsl/fsl_spdif.c | 73 +- sound/soc/fsl/fsl_spdif.h | 9 +- sound/soc/fsl/fsl_xcvr.c | 1360 +++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++ sound/soc/fsl/imx-audmux.c | 28 +- sound/soc/fsl/imx-hdmi.c | 236 ++ sound/soc/fsl/imx-mc13783.c | 156 - sound/soc/fsl/imx-ssi.c | 651 ---- sound/soc/fsl/mx27vis-aic32x4.c | 214 -- sound/soc/fsl/phycore-ac97.c | 121 - sound/soc/fsl/wm1133-ev1.c | 289 -- sound/soc/generic/Kconfig | 2 +- sound/soc/generic/audio-graph-card.c | 175 +- sound/soc/intel/Kconfig | 5 +- sound/soc/intel/atom/sst-atom-controls.c | 12 +- sound/soc/intel/atom/sst-atom-controls.h | 4 +- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 8 + sound/soc/intel/boards/Kconfig | 3 +- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/bdw-rt5650.c | 17 +- sound/soc/intel/boards/bdw-rt5677.c | 18 +- sound/soc/intel/boards/broadwell.c | 20 +- sound/soc/intel/boards/bytcht_cx2072x.c | 27 +- sound/soc/intel/boards/bytcht_da7213.c | 27 +- sound/soc/intel/boards/bytcht_es8316.c | 29 +- sound/soc/intel/boards/bytcr_rt5640.c | 42 +- sound/soc/intel/boards/bytcr_rt5651.c | 29 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 29 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 39 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 38 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 29 +- sound/soc/intel/boards/sof_maxim_common.c | 4 +- sound/soc/intel/boards/sof_realtek_common.c | 138 + sound/soc/intel/boards/sof_realtek_common.h | 24 + sound/soc/intel/boards/sof_rt5682.c | 87 +- sound/soc/intel/boards/sof_sdw.c | 10 + sound/soc/intel/catpt/core.h | 10 +- sound/soc/intel/catpt/device.c | 30 +- sound/soc/intel/catpt/dsp.c | 56 +- sound/soc/intel/catpt/loader.c | 2 +- sound/soc/intel/catpt/pcm.c | 106 +- sound/soc/intel/common/Makefile | 2 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 52 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 13 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 17 +- sound/soc/intel/keembay/kmb_platform.c | 22 +- sound/soc/intel/keembay/kmb_platform.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/Kconfig | 25 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 1 - sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 +++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 ++ sound/soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 308 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1471 +++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2110 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 +++++ sound/soc/mediatek/mt8192/mt8192-interconnection.h | 65 + .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 1137 +++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 ++++++++++++++++++++ sound/soc/meson/Kconfig | 4 +- sound/soc/meson/t9015.c | 2 +- sound/soc/pxa/Kconfig | 14 +- sound/soc/pxa/mmp-sspa.c | 4 + sound/soc/pxa/pxa-ssp.c | 6 +- sound/soc/qcom/Kconfig | 25 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/common.c | 13 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-cpu.c | 12 +- sound/soc/qcom/lpass-hdmi.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 10 +- sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/qdsp6/q6adm.c | 10 +- sound/soc/qcom/qdsp6/q6afe-clocks.c | 3 +- sound/soc/qcom/qdsp6/q6afe.c | 10 +- sound/soc/qcom/qdsp6/q6asm.c | 10 +- sound/soc/qcom/sc7180.c | 391 +++ sound/soc/qcom/sm8250.c | 229 ++ sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 11 +- sound/soc/samsung/midas_wm1811.c | 1 - sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/snow.c | 2 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-component.c | 318 +- sound/soc/soc-compress.c | 397 +-- sound/soc/soc-core.c | 90 +- sound/soc/soc-dai.c | 70 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-link.c | 53 +- sound/soc/soc-pcm.c | 217 +- sound/soc/soc-topology.c | 337 +-- sound/soc/sof/Kconfig | 42 +- sound/soc/sof/control.c | 42 +- sound/soc/sof/debug.c | 119 +- sound/soc/sof/imx/Kconfig | 10 +- sound/soc/sof/imx/imx-common.c | 2 + sound/soc/sof/intel/Kconfig | 110 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/apl.c | 3 + sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/cnl.c | 19 +- sound/soc/sof/intel/ext_manifest.h | 35 + sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-loader.c | 119 +- sound/soc/sof/intel/hda-pcm.c | 7 + sound/soc/sof/intel/hda.c | 10 +- sound/soc/sof/intel/hda.h | 12 + sound/soc/sof/intel/icl.c | 145 + sound/soc/sof/intel/intel-ipc.c | 7 + sound/soc/sof/intel/tgl.c | 19 + sound/soc/sof/ipc.c | 9 + sound/soc/sof/loader.c | 62 +- sound/soc/sof/nocodec.c | 18 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 14 +- sound/soc/sof/pcm.c | 14 +- sound/soc/sof/sof-acpi-dev.c | 14 +- sound/soc/sof/sof-audio.c | 12 +- sound/soc/sof/sof-audio.h | 5 + sound/soc/sof/sof-pci-dev.c | 24 +- sound/soc/sof/sof-priv.h | 22 +- sound/soc/sof/topology.c | 33 +- sound/soc/sof/trace.c | 224 ++ sound/soc/stm/Kconfig | 1 + sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 388 ++- sound/soc/sunxi/sun8i-codec.c | 834 +++++- sound/soc/tegra/Kconfig | 1 - sound/soc/tegra/tegra186_dspk.c | 6 +- sound/soc/tegra/tegra210_dmic.c | 6 +- sound/soc/tegra/tegra210_i2s.c | 6 +- sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_max98090.c | 8 +- sound/soc/tegra/tegra_rt5640.c | 8 +- sound/soc/tegra/tegra_rt5677.c | 4 +- sound/soc/tegra/tegra_sgtl5000.c | 4 +- sound/soc/tegra/tegra_wm8753.c | 8 +- sound/soc/tegra/tegra_wm8903.c | 8 +- sound/soc/tegra/tegra_wm9712.c | 4 +- sound/soc/tegra/trimslice.c | 8 +- sound/soc/ti/Kconfig | 9 +- sound/soc/ti/davinci-evm.c | 3 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 298 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/usb/pcm.c | 6 + sound/usb/quirks.c | 1 + tools/arch/arm64/include/uapi/asm/kvm.h | 25 + tools/arch/s390/include/uapi/asm/sie.h | 2 +- tools/arch/x86/include/asm/cpufeatures.h | 6 +- tools/arch/x86/include/asm/disabled-features.h | 9 +- tools/arch/x86/include/asm/msr-index.h | 10 + tools/arch/x86/include/asm/required-features.h | 2 +- tools/arch/x86/include/uapi/asm/kvm.h | 20 + tools/arch/x86/include/uapi/asm/svm.h | 13 + tools/build/feature/test-all.c | 1 - tools/include/linux/compiler-gcc.h | 12 - tools/include/linux/compiler.h | 3 - tools/include/uapi/asm-generic/unistd.h | 4 +- tools/include/uapi/drm/i915_drm.h | 59 +- tools/include/uapi/linux/fscrypt.h | 6 +- tools/include/uapi/linux/kvm.h | 19 + tools/include/uapi/linux/mman.h | 1 + tools/include/uapi/linux/mount.h | 1 + tools/include/uapi/linux/perf_event.h | 2 +- tools/include/uapi/linux/prctl.h | 9 + tools/include/uapi/linux/vhost.h | 4 + tools/perf/Makefile.config | 1 + 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mode 100644 include/linux/scif.h create mode 100644 include/sound/graph_card.h create mode 100644 include/sound/soc-jack.h create mode 100644 include/sound/sof/debug.h delete mode 100644 include/uapi/linux/mic_common.h delete mode 100644 include/uapi/linux/mic_ioctl.h delete mode 100644 samples/mic/mpssd/.gitignore delete mode 100644 samples/mic/mpssd/Makefile delete mode 100755 samples/mic/mpssd/micctrl delete mode 100755 samples/mic/mpssd/mpss delete mode 100644 samples/mic/mpssd/mpssd.c delete mode 100644 samples/mic/mpssd/mpssd.h delete mode 100644 samples/mic/mpssd/sysfs.c create mode 100644 sound/soc/codecs/adau1372-i2c.c create mode 100644 sound/soc/codecs/adau1372-spi.c create mode 100644 sound/soc/codecs/adau1372.c create mode 100644 sound/soc/codecs/adau1372.h create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h create mode 100644 sound/soc/codecs/nau8315.c create mode 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sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h create mode 100644 sound/soc/qcom/sc7180.c create mode 100644 sound/soc/qcom/sm8250.c create mode 100644 sound/soc/sof/intel/ext_manifest.h create mode 100644 sound/soc/sof/intel/icl.c create mode 100644 tools/testing/selftests/kvm/x86_64/vmx_apic_access_test.c

2 1

[RFC][PATCH 0/2] design a way to change audio Jack state by software
by Hui Wang 14 Dec '20

14 Dec '20

After we change sth in the userspace audio stack like alsa-ucm or pulseaudio, we want to perform remote audio auto test to verify if the change introduce the regression or not, some of the tests are about the defaut_sink/default_source or active_port switching, this needs the audio jack state to be changed to trigger the userspace's audio device switching. So far, there is no software ways to change the audio jack state, this block the auto test. My design is adding a sysfs interface for each sound card if the card has audio jack, then users could echo different values to sysfs to change the jack state (Phantom jack is not controlled by injection). And once the users enable the jack injection via sysfs, this jack's state will not be controlled by hw events anymore until users disable the jack injection. Of course, this could not 100% simulate the plugin or plugout triggered by hw events, with the hw triggered plugin or plugout, the audio driver will set codec or does sth else, so the software injection is just changing the jack state and notify the userspace, it is just for testing userspace part. Here is an example to change jack state via sysfs: After booting up: /* cd to the jack injection folder for sound card0 in the sysfs */ $cd /sys/devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0/jack /* check file nodes in this folder */ $ls jackin_inject sw_inject_enable /* check all jack's software injection enable status, all disabled now */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 0 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 /* enable software injection for Jack Headphone */ $ sudo sh -c "echo Headphone 1 > sw_inject_enable" /* check all jack's software injection enable status again, now Headphone is enabled */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 1 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 /* trigger plugin to Jack Headphone */ $sudo sh -c "echo Headphone 1 > jackin_inject" /* check if Jack Headphone is plugged in */ $ sudo amixer contents | grep "Headphone Jack" -3 numid=30,iface=CARD,name='HDMI/DP,pcm=5 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=17,iface=CARD,name='Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=14,iface=CARD,name='Mic Jack' /* trigger plugout to Jack Headphone */ $ sudo sh -c "echo Headphone 0 > jackin_inject" /* check if Jack Headphone is plugged out */ $ sudo amixer contents | grep "Headphone Jack" -3 numid=30,iface=CARD,name='HDMI/DP,pcm=5 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=17,iface=CARD,name='Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=14,iface=CARD,name='Mic Jack' /* disable Jack Headphone software injection, this will return the control to non-injection ways */ $ sudo sh -c "echo Headphone 0 > sw_inject_enable" /* check if the Jack Headphone software injection is disabled, it is disabled now */ $ cat sw_inject_enable Jack: Mic 0 Jack: Headphone 0 Jack: HDMI/DP,pcm=3 0 Jack: HDMI/DP,pcm=4 0 Jack: HDMI/DP,pcm=5 0 Hui Wang (2): alsa: jack: expand snd_jack_report parameter for jack sw_inject alsa: jack: adding support for software jack in or out injection include/sound/core.h | 1 + include/sound/jack.h | 5 +- sound/core/jack.c | 129 +++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.c | 6 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 2 +- sound/x86/intel_hdmi_audio.c | 4 +- 8 files changed, 140 insertions(+), 11 deletions(-) -- 2.25.1

4 7

[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 14 Dec '20

14 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 14 Dec '20

14 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH alsa-lib 1/3] chtnau8824: Fix mono speaker config not working
by Hans de Goede 14 Dec '20

14 Dec '20

Fix the Speaker define / var being changed from "Speaker" to "MonoSpeaker" not having any effect because the variable gets expanded before the if making the change is interpreted. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/chtnau8824/HiFi.conf | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 3ccd6b1..d7a5f63 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -34,8 +34,15 @@ SectionVerb { } } -Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" -Include.hp.File "/codecs/nau8824/HeadPhones.conf" +# The includes using $vars in there path must be conditional otherwise the +# $var gets expanded before the other If-s above can change the vars. +If.cfg-includes { + Condition { Type String Empty "" } + True { + Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" + Include.hp.File "/codecs/nau8824/HeadPhones.conf" -Include.mic.File "/codecs/nau8824/InternalMic.conf" -Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + Include.mic.File "/codecs/nau8824/InternalMic.conf" + Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + } +} -- 2.28.0

2 4

[RFC PATCH 0/4] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 14 Dec '20

14 Dec '20

v1: Initial patch version, to enable DMA mode on Intel Keembay platform by exposing some dmaengine api to work around DMA limitations at the client driver level. This patchset suggests an ALSA-only quirk, having other more generic means to deal with this limitation would be fine - we just wanted to have a discussion on preferred directions. The IPs used are not Intel-specific so sooner or later someone else will have similar limitations to work-around. Michael Sit Wei Hong (4): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: soc-generic-dmaengine-pcm: Add custom prepare and submit function ASoC: dmaengine_pcm: expose functions to header file for custom functions ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ include/sound/dmaengine_pcm.h | 21 ++ sound/core/pcm_dmaengine.c | 46 ++-- sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 202 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + sound/soc/soc-generic-dmaengine-pcm.c | 8 +- 7 files changed, 267 insertions(+), 35 deletions(-) -- 2.17.1

8 22

Problem with synthesized output
by Spencer Kohan 13 Dec '20

13 Dec '20

Hello, I hope I am posting this to the correct place. I'm trying to use a Raspberry Pi as a wifi speaker, and as a first step I'm trying to just play some sound out of a buffer through the attached amp. I'm using an AMP2 with a RPi 4, and I'm getting some strange behavior when I try to play any "sine" output. So for instance, if I use `speaker-test`, I am able to play the wav test and the pink-noise test just fine, but when I try to play the sine test, it plays for like one quick instant, and then the program blocks. What's strange is I get the exact same behavior when I use other methods to try to play a sine output over ASLA. For instance, if I use sox: play -n synth sine 1000 I get the same output: one quick blip of the tone, and then the program blocks. The output to the console looks like this: play WARN alsa: can't encode 0-bit Unknown or not applicable File Size: 0 Encoding: n/a Channels: 1 @ 32-bit Samplerate: 48000Hz Replaygain: off Duration: unknown In:0.00% 00:00:00.00 [00:00:00.00] Out:0 [ | ] Clip:0 And then I get the exact same behavior when I try to do the `sio_sine` example from [libsoundio](https://github.com/andrewrk/libsoundio), with this output: Backend: ALSA ALSA lib pcm_dsnoop.c:575:(snd_pcm_dsnoop_open) The dsnoop plugin supports only capture stream Output device: snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0: Default Audio Device Software latency: 0.008000 'p\n' - pause 'u\n' - unpause 'P\n' - pause from within callback 'c\n' - clear buffer 'q\n' - quit unable to set channel layout: incompatible device Also interesting, when I try the [rsoundio]( https://github.com/klingtnet/rsoundio) sine example, it reports an underflow. My output for aplay looks like this: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device sysdefault:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device dmix:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample mixing device dsnoop:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample snooping device hw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct hardware device without any conversions plughw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Hardware device with all software conversions It's just strange because the pre-recorded sample and the pink noise version work just fine. How would I even go about trying to debug this?

1 0

Bug in the ALSA driver 'snd_usb_audio'
by GitHub issues - opened 12 Dec '20

12 Dec '20

alsa-project/alsa-lib issue #105 was opened from rombru: Hello, I'm not an expert in linux systems. I will try to give a maximum of information. I've bought Bowers & Wilkins PX7 headphones. They doesn't work using USB on Ubuntu. Using Ubuntu and bluetooth, or audio jack, the headphones work well. Using Windows 10 and USB, the headphones work well. I d'like too use them with USB on Ubuntu because that recharge them in parallel. Problems while the headphons are connected : On Chrome : The videos are played too fast and there's no sound (Youtube & Twitch). On Firefox : The videos are played too fast and there's a crackling sound (Youtube & Twitch). On VLC : No sound I've tried with Ubuntu 20.04 & 18.04. They are set as default audio device I've played with PulseAudio and alsamixer (reinstall, mute/unmute, ...) The headphones are set as fallback in PulseAudio. Trying to find the issue, i've make some test found on Google : I've added pcie_aspm=off in GRUB_CMDLINE_LINUX_DEFAULT Setting devices as default in /etc/asound.conf --- The ALSA part I think the problem come from ALSA, i've found the errors : `Failed to load module "module-alsa-card` followed few lines later by `Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers.` Output of `journalctl --user -u pulseaudio` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pulseaudio -vvvvv` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pacmd ls` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Please, tell me if you need something more to analyse the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/105 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 5.9 06/23] ASoC: rt5682: change SAR voltage threshold
by Sasha Levin 12 Dec '20

12 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit aa4cb898b80a28a610e26d1513e6dd42d995c225 ] To fix errors in some 4 poles headset detection cases, this patch adjusts the voltage threshold for mic detection. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20201126092759.9427-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a4713bd6508d5..e260eddb51d96 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, + {RT5682_SAR_IL_CMD_3, 0x8365}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.27.0

1 0

[PATCH 1/2] ALSA: hda/ca0132 - Add 8051 PLL write helper functions.
by Connor McAdams 12 Dec '20

12 Dec '20

Add helper functions for the 8051 PLL PMU write verbs. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 130 ++++++++++++++--------------------- 1 file changed, 50 insertions(+), 80 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e9019e840a4d..9b6b7601a881 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1949,6 +1949,14 @@ static unsigned int chipio_8051_get_data(struct hda_codec *codec) VENDOR_CHIPIO_8051_DATA_READ, 0); } +/* PLL_PMU writes share the lower address register of the 8051 exram writes. */ +static void chipio_8051_set_data_pll(struct hda_codec *codec, unsigned int data) +{ + /* 8-bits of data. */ + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, data & 0xff); +} + static void chipio_8051_write_exram(struct hda_codec *codec, unsigned int addr, unsigned int data) { @@ -1977,6 +1985,26 @@ static void chipio_8051_read_exram(struct hda_codec *codec, *data = chipio_8051_get_data(codec); } +static void chipio_8051_write_pll_pmu(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); + + mutex_unlock(&spec->chipio_mutex); +} + +static void chipio_8051_write_pll_pmu_no_mutex(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); +} + /* * Enable clocks. */ @@ -1985,18 +2013,11 @@ static void chipio_enable_clocks(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 5); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x0b); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 6); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); + + chipio_8051_write_pll_pmu_no_mutex(codec, 0x00, 0xff); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x05, 0x0b); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x06, 0xff); + mutex_unlock(&spec->chipio_mutex); } @@ -7904,10 +7925,7 @@ static void ae5_post_dsp_register_set(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); writeb(0xff, spec->mem_base + 0x304); writeb(0xff, spec->mem_base + 0x304); @@ -7949,30 +7967,11 @@ static void ae5_post_dsp_param_setup(struct hda_codec *codec) static void ae5_post_dsp_pll_setup(struct hda_codec *codec) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d); + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); + chipio_8051_write_pll_pmu(codec, 0x45, 0xcc); + chipio_8051_write_pll_pmu(codec, 0x40, 0xcb); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x51, 0x8d); } static void ae5_post_dsp_stream_setup(struct hda_codec *codec) @@ -7994,10 +7993,7 @@ static void ae5_post_dsp_stream_setup(struct hda_codec *codec) chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80); @@ -8090,12 +8086,8 @@ static void ae7_post_dsp_pll_setup(struct hda_codec *codec) }; unsigned int i; - for (i = 0; i < ARRAY_SIZE(addr); i++) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]); - } + for (i = 0; i < ARRAY_SIZE(addr); i++) + chipio_8051_write_pll_pmu_no_mutex(codec, addr[i], data[i]); } static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) @@ -8111,10 +8103,7 @@ static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); chipio_write_no_mutex(codec, 0x189000, 0x0001f101); chipio_write_no_mutex(codec, 0x189004, 0x0001f101); @@ -8187,10 +8176,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) { chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); @@ -8207,10 +8193,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) ae7_post_dsp_pll_setup(codec); ae7_post_dsp_asi_stream_setup(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); ae7_post_dsp_asi_setup_ports(codec); } @@ -9367,18 +9350,11 @@ static void ae5_register_set(struct hda_codec *codec) unsigned int i, cur_addr; unsigned char tmp[3]; - if (ca0132_quirk(spec) == QUIRK_AE7) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - } + if (ca0132_quirk(spec) == QUIRK_AE7) + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); if (ca0132_quirk(spec) == QUIRK_AE7) { tmp[0] = 0x03; @@ -9454,10 +9430,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE5: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); chipio_write(codec, 0x18b030, 0x00000020); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); @@ -9465,10 +9438,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE7: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); chipio_write(codec, 0x18b008, 0x000000f8); -- 2.25.1

2 2

[PATCH alsa-utils] alsactl: Remove asound.state file check from alsa-restore.service again
by Takashi Iwai 11 Dec '20

11 Dec '20

We added the check of asound.state file presence some time ago to assure that alsactl gets called only if the state file is already present. Since then, the situation has changed significantly: e.g. now alsactl does initialize if the state file isn't present, and the same alsa-restore.service is used to save the state. This means that we should start this service no matter the state file exists at the boot time or not. So, revert the old change again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/alsa-restore.service.in | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index a84c2e842444..80fd5fd48203 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -7,7 +7,6 @@ Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -ConditionPathExists=@asoundrcfile@ [Service] Type=oneshot -- 2.26.2

1 0

[PATCH alsa-utils 1/2] alsactl: Fix double decrease of lock timeout
by Takashi Iwai 11 Dec '20

11 Dec '20

The state_lock() has a loop to wait for the lock file creation, and the timeout value gets decremented twice mistakenly, which leads to a half timeout (5 seconds) than expected 10 seconds. Fix it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/lock.c | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index 4a485392b3bd..05f6e4d2a102 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -63,7 +63,6 @@ static int state_lock_(const char *file, int lock, int timeout, int _fd) if (fd < 0) { if (errno == EBUSY || errno == EAGAIN) { sleep(1); - timeout--; } else { err = -errno; goto out; -- 2.26.2

1 1

[PATCH 1/2] dt-bindings: add simple-audio-mux binding
by Alexandre Belloni 11 Dec '20

11 Dec '20

Add devicetree documentation for simple audio multiplexers Signed-off-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> --- Cc: Rob Herring <robh+dt(a)kernel.org> .../bindings/sound/simple-audio-mux.yaml | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-mux.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000000..5986d1fcbb54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Multiplexer + +maintainers: + - Alexandre Belloni <aleandre.belloni(a)bootlin.com> + +description: | + Simple audio multiplexers are driven using gpios, allowing to select which of + their input line is connected to the output line. + +properties: + compatible: + const: simple-audio-mux + + mux-gpios: + description: | + GPIOs used to select the input line. + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: + Used as prefix for sink/source names of the component. Must be a + unique string among multiple instances of the same component. + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + }; -- 2.28.0

3 4

[PATCH] alsactl: Skip restore during the lock
by Takashi Iwai 11 Dec '20

11 Dec '20

Currently alsactl-restore tries to initialize the device when an error is found for restore action. But this isn't the right behavior in the case where the lock is held; it implies that another alsactl is running concurrently, hence you shouldn't initialize the card at the same time. The situation is found easily when two alsactls get started by both udev and systemd (note that those two invocations are the designed behavior, see /usr/lib/udev/rules.d/78-sound-cards.rules for details). This patch changes load_state() not to handle the initialization if the locking fails. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1179904 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/state.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/alsactl/state.c b/alsactl/state.c index ea1d3bcaaddf..8d1d814341bd 100644 --- a/alsactl/state.c +++ b/alsactl/state.c @@ -1663,7 +1663,12 @@ int load_state(const char *file, const char *initfile, int initflags, err = snd_input_stdio_attach(&in, stdin, 0); } else { lock_fd = state_lock(file, 10); - err = lock_fd >= 0 ? snd_input_stdio_open(&in, file, "r") : lock_fd; + /* skip restore if already locked; i.e. concurrent accesses */ + if (lock_fd < 0) { + err = 0; + goto out_global; + } + err = snd_input_stdio_open(&in, file, "r"); } if (err >= 0) { err = snd_config_load(config, in); @@ -1781,6 +1786,7 @@ single: err = finalerr; out: snd_config_delete(config); +out_global: snd_config_update_free_global(); return err; } -- 2.26.2

2 10

[PATCH] ASoC: SOF: imx: update kernel-doc description
by Kai Vehmanen 11 Dec '20

11 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add missing parameters to avoid W=1 error Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/imx/imx-common.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 5fee637834c2..8826ef94f04a 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -47,6 +47,8 @@ void imx8_get_registers(struct snd_sof_dev *sdev, /** * imx8_dump() - This function is called when a panic message is * received from the firmware. + * @sdev: SOF device + * @flags: parameter not used but required by ops prototype */ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) { -- 2.29.2

2 1

[PATCH 0/3] ASoC: SOF: Intel: fix to dsp state dump trace levels
by Kai Vehmanen 11 Dec '20

11 Dec '20

Small series that addresses a problem where DSP status dump for a failure case, ends up being printed as as debug print. This is important information for any bug report. While at it, the series contains a few cleanups to related code. Ranjani Sridharan (3): ASoC: SOF: Intel: hda: remove duplicated status dump ASoC: SOF: modify the SOF_DBG flags ASoC: SOF: Intel: hda: fix the condition passed to sof_dev_dbg_or_err sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/hda-loader.c | 19 +++++++++++++------ sound/soc/sof/intel/hda.c | 10 ++++------ sound/soc/sof/loader.c | 4 ++-- sound/soc/sof/ops.c | 2 +- sound/soc/sof/sof-priv.h | 13 ++++++++----- 7 files changed, 30 insertions(+), 22 deletions(-) -- 2.29.2

2 4

[PATCH] ASoC: mediatek: mt8183: delete some unreachable code
by Dan Carpenter 11 Dec '20

11 Dec '20

This has a goto followed by an unreachable return statement. The goto is correct because it cleans up so the current runtime behavior is fine. Let's delete the unreachable return statement. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-clk.c b/sound/soc/mediatek/mt8183/mt8183-afe-clk.c index 48e81c5d52fc..cc4f8f4d3dab 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-clk.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-clk.c @@ -584,7 +584,6 @@ int mt8183_mck_enable(struct mtk_base_afe *afe, int mck_id, int rate) __func__, aud_clks[div_clk_id], rate, ret); goto ERR_SET_MCLK_RATE; - return ret; } return 0; -- 2.29.2

2 1

[PATCH] ASoC: mediatek: mt8183: add PM ops to machine drivers
by Tzung-Bi Shih 11 Dec '20

11 Dec '20

Adds PM ops to machine drivers so that they notify components in the sound card when system suspend. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 26e7d9a7198f..078e58f1ad0b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -811,6 +811,7 @@ static struct platform_driver mt8183_da7219_max98357_driver = { #ifdef CONFIG_OF .of_match_table = mt8183_da7219_max98357_dt_match, #endif + .pm = &snd_soc_pm_ops, }, .probe = mt8183_da7219_max98357_dev_probe, }; diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 327dfad41e31..8c8340854859 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -744,6 +744,7 @@ static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { #ifdef CONFIG_OF .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, #endif + .pm = &snd_soc_pm_ops, }, .probe = mt8183_mt6358_ts3a227_max98357_dev_probe, }; -- 2.29.2.684.gfbc64c5ab5-goog

2 1

[PATCH v3 0/2] ASoC: rt1015p: delay 300ms for waiting calibration
by Tzung-Bi Shih 11 Dec '20

11 Dec '20

The 1st patch moves SDB control from DAI ops trigger to DAPM event (per review comments in v1). The 2nd patch adds the 300ms delay for waiting calibration. Changes from v2: - Use gpiod_set_value_cansleep() instead of gpiod_set_value(). (https://patchwork.kernel.org/project/alsa-devel/patch/20201210033617.79300-…) - Assuming the calibration state gets lost after system suspend. (https://patchwork.kernel.org/project/alsa-devel/patch/20201210033617.79300-…) Changes from v1: (https://patchwork.kernel.org/project/alsa-devel/patch/20201209033742.382597…) - Move the delay from trigger to DAPM event. Tzung-Bi Shih (2): ASoC: rt1015p: move SDB control from trigger to DAPM ASoC: rt1015p: delay 300ms after SDB pulling high for calibration sound/soc/codecs/rt1015p.c | 69 ++++++++++++++++++-------------------- 1 file changed, 32 insertions(+), 37 deletions(-) -- 2.29.2.684.gfbc64c5ab5-goog

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[PATCH 1/2] ASoC: topology: Fix wrong size check
by Amadeusz Sławiński 11 Dec '20

11 Dec '20

Dan reported that smatch reports wrong size check and after analysis it is confirmed that we are comparing wrong value: pointer size instead of array size. However the check itself is problematic as in UAPI header there are two fields: struct snd_soc_tplg_enum_control { (...) char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; the texts field is for names and the values one for values assigned to those named fields, after analysis it becomes clear that there is quite a lot overhead values than we may possibly name. So instead of changing check to ARRAY_SIZE(ec->values), as it was first suggested, use hardcoded value of SND_SOC_TPLG_NUM_TEXTS. Link: https://lore.kernel.org/alsa-devel/X9B0eDcKy+9B6kZl@mwanda/ Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index eb2633dd6454..7fb3a87ab860 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -889,10 +889,16 @@ static int soc_tplg_denum_create_values(struct soc_tplg *tplg, struct soc_enum * { int i; - if (le32_to_cpu(ec->items) > sizeof(*ec->values)) + /* + * Following "if" checks if we have at most SND_SOC_TPLG_NUM_TEXTS + * values instead of using ARRAY_SIZE(ec->values) due to the fact that + * it is oversized for its purpose. Additionally it is done so because + * it is defined in UAPI header where it can't be easily changed. + */ + if (le32_to_cpu(ec->items) > SND_SOC_TPLG_NUM_TEXTS) return -EINVAL; - se->dobj.control.dvalues = devm_kzalloc(tplg->dev, le32_to_cpu(ec->items) * + se->dobj.control.dvalues = devm_kcalloc(tplg->dev, le32_to_cpu(ec->items), sizeof(u32), GFP_KERNEL); if (!se->dobj.control.dvalues) -- 2.25.1

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[PATCH] ASoC: wm_adsp: remove "ctl" from list on error in wm_adsp_create_control()
by Dan Carpenter 11 Dec '20

11 Dec '20

The error handling frees "ctl" but it's still on the "dsp->ctl_list" list so that could result in a use after free. Remove it from the list before returning. Fixes: 2323736dca72 ("ASoC: wm_adsp: Add basic support for rev 1 firmware file format") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e61d00486c65..dec8716aa8ef 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1519,7 +1519,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); if (!ctl_work) { ret = -ENOMEM; - goto err_ctl_cache; + goto err_list_del; } ctl_work->dsp = dsp; @@ -1529,7 +1529,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, return 0; -err_ctl_cache: +err_list_del: + list_del(&ctl->list); kfree(ctl->cache); err_ctl_subname: kfree(ctl->subname); -- 2.29.2

3 2

[RESEND PATCH 0/2] Convert Tegra HDA doc to json-schema
by Sameer Pujar 11 Dec '20

11 Dec '20

Convert Tegra HDA doc to YAML format. Ran binding validation checks to ensure there are no warnings/errors for HDA. Patch "arm64: tegra: Fix Tegra194 HDA {clock,reset}-names ordering" is already merged and hence not included here. Sameer Pujar (2): dt-bindings: tegra: Convert HDA doc to json-schema dt-bindings: tegra: Add missing HDA properties .../bindings/sound/nvidia,tegra30-hda.txt | 35 ------- .../bindings/sound/nvidia,tegra30-hda.yaml | 112 +++++++++++++++++++++ 2 files changed, 112 insertions(+), 35 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml -- 2.7.4

3 5

[PATCH] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Kai Vehmanen 11 Dec '20

11 Dec '20

The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2ddc27db8c01..81fd7f9a40ce 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -638,11 +638,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.29.2

2 2

[PATCH] ALSA: hda/hdmi: always print pin NIDs as hexadecimal
by Kai Vehmanen 11 Dec '20

11 Dec '20

The debug prints from patch_hdmi.c are not aligned with HDA common code in hda_codec.c nor with other HDA codec drivers. To align with rest of the codebase, use hexadecimal formatting whenever printing value of a HDA NID. Also refer to NIDs with capital letters in traces as is done other modules. This presentation is also aligned with the formatting used in HDA codec procfs entry. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 27 ++++++++++++--------------- 1 file changed, 12 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2ddc27db8c01..22f5a29d25f6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -253,7 +253,7 @@ static int pin_id_to_pin_index(struct hda_codec *codec, return pin_idx; } - codec_warn(codec, "HDMI: pin nid %d not registered\n", pin_nid); + codec_warn(codec, "HDMI: pin NID 0x%x not registered\n", pin_nid); return -EINVAL; } @@ -313,7 +313,7 @@ static int cvt_nid_to_cvt_index(struct hda_codec *codec, hda_nid_t cvt_nid) if (get_cvt(spec, cvt_idx)->cvt_nid == cvt_nid) return cvt_idx; - codec_warn(codec, "HDMI: cvt nid %d not registered\n", cvt_nid); + codec_warn(codec, "HDMI: cvt NID 0x%x not registered\n", cvt_nid); return -EINVAL; } @@ -687,8 +687,7 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, dp_ai->CC02_CT47 = active_channels - 1; dp_ai->CA = ca; } else { - codec_dbg(codec, "HDMI: unknown connection type at pin %d\n", - pin_nid); + codec_dbg(codec, "HDMI: unknown connection type at pin NID 0x%x\n", pin_nid); return; } @@ -701,10 +700,8 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, */ if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { - codec_dbg(codec, - "hdmi_pin_setup_infoframe: pin=%d channels=%d ca=0x%02x\n", - pin_nid, - active_channels, ca); + codec_dbg(codec, "%s: pin NID=0x%x channels=%d ca=0x%02x\n", + __func__, pin_nid, active_channels, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -796,7 +793,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res, jack->jack_dirty = 1; codec_dbg(codec, - "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + "HDMI hot plug event: Codec=%d NID=0x%x Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, jack->nid, jack->dev_id, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); @@ -874,7 +871,7 @@ static void haswell_verify_D0(struct hda_codec *codec, msleep(40); pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; - codec_dbg(codec, "Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); + codec_dbg(codec, "Haswell HDMI audio: Power for NID 0x%x is now D%d\n", nid, pwr); } } @@ -1121,8 +1118,8 @@ static void intel_not_share_assigned_cvt(struct hda_codec *codec, per_cvt = get_cvt(spec, cvt_idx); if (!per_cvt->assigned) { codec_dbg(codec, - "choose cvt %d for pin nid %d\n", - cvt_idx, nid); + "choose cvt %d for pin NID 0x%x\n", + cvt_idx, nid); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, cvt_idx); @@ -1320,7 +1317,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { codec_warn(codec, - "HDMI: pin %d wcaps %#x does not support connection list\n", + "HDMI: pin NID 0x%x wcaps %#x does not support connection list\n", pin_nid, get_wcaps(codec, pin_nid)); return -EINVAL; } @@ -1635,7 +1632,7 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, eld->eld_valid = false; codec_dbg(codec, - "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + "HDMI status: Codec=%d NID=0x%x Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { @@ -2795,7 +2792,7 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) return i; } - codec_info(codec, "Can't find the HDMI/DP port for pin %d\n", pin_nid); + codec_info(codec, "Can't find the HDMI/DP port for pin NID 0x%x\n", pin_nid); return -1; } -- 2.29.2

2 1

[PATCH] ALSA: usb-audio: Fix control 'access overflow' errors from chmap
by Takashi Iwai 11 Dec '20

11 Dec '20

The current channel-map control implementation in USB-audio driver may lead to an error message like "control 3:0:0:Playback Channel Map:0: access overflow" when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get callback clears the whole array no matter which count is set, and rather the false-positive detection. This patch fixes the problem by clearing only the needed array range at usb_chmap_ctl_get(). Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ca76ba5b5c0b..2f6d39c2ba7c 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -193,16 +193,16 @@ static int usb_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct snd_usb_substream *subs = info->private_data; struct snd_pcm_chmap_elem *chmap = NULL; - int i; + int i = 0; - memset(ucontrol->value.integer.value, 0, - sizeof(ucontrol->value.integer.value)); if (subs->cur_audiofmt) chmap = subs->cur_audiofmt->chmap; if (chmap) { for (i = 0; i < chmap->channels; i++) ucontrol->value.integer.value[i] = chmap->map[i]; } + for (; i < subs->channels_max; i++) + ucontrol->value.integer.value[i] = 0; return 0; } -- 2.26.2

1 0

How to annotate USB jacks without jack detection?
by Kai-Heng Feng 11 Dec '20

11 Dec '20

Hi, There are some UAC1 devices come with jacks, but without jack detection ability. And many USB/Thunderbolt docks are equipped with such USB device. Userspace like PulseAudio automatically switch to USB audio when they gets plugged. However, auto-switching to UAC1 device can be problematic because it may not connected to any headset or speaker. So I wonder what's the best way to let userspace know the USB device is a jack without jack detection ability? Through sysfs or dedicated UCM conf? Kai-Heng

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[PATCH] ASoC: fsl: imx-hdmi: Fix an uninitialized return in probe
by Dan Carpenter 11 Dec '20

11 Dec '20

The "ret" variable should be set to a negative error code but currently it returns uninitialized stack data. Fixes: 6a5f850aa83a ("ASoC: fsl: Add imx-hdmi machine driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.29.2

1 0

Add supported for more Lenovo ALC285 Headset Button -- again
by Kailang 11 Dec '20

11 Dec '20

Hi Takashi, This platform had more SSID. So, use pin quirk for this platform. Many Thanks. Kailang

2 1