From 0bd8bce897b6697bbc286b8ba473aa0705fe394b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 23 Aug 2020 16:55:37 +0900 Subject: ALSA; firewire-tascam: exclude Tascam FE-8 from detection Tascam FE-8 is known to support communication by asynchronous transaction only. The support can be implemented in userspace application and snd-firewire-ctl-services project has the support. However, ALSA firewire-tascam driver is bound to the model. This commit changes device entries so that the model is excluded. In a commit 53b3ffee7885 ("ALSA: firewire-tascam: change device probing processing"), I addressed to the concern that version field in configuration differs depending on installed firmware. However, as long as I checked, the version number is fixed. It's safe to return version number back to modalias. Fixes: 53b3ffee7885 ("ALSA: firewire-tascam: change device probing processing") Cc: # 4.4+ Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200823075537.56255-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 5dac0d9fc58e..75f2edd8e78f 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, - // This kernel module doesn't support FE-8 because the most of features - // can be implemented in userspace without any specific support of this - // module. }; static int identify_model(struct snd_tscm *tscm) @@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit) } static const struct ieee1394_device_id snd_tscm_id_table[] = { + // Tascam, FW-1884. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800000, + }, + // Tascam, FE-8 (.version = 0x800001) + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. + // + // .version = 0x800002 is unknown. + // + // Tascam, FW-1082. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800003, + }, + // Tascam, FW-1804. { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, .vendor_id = 0x00022e, .specifier_id = 0x00022e, + .version = 0x800004, }, {} }; -- cgit v1.2.1 From acd46a6b6de88569654567810acad2b0a0a25cea Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 23 Aug 2020 16:55:45 +0900 Subject: ALSA: firewire-digi00x: exclude Avid Adrenaline from detection Avid Adrenaline is reported that ALSA firewire-digi00x driver is bound to. However, as long as he investigated, the design of this model is hardly similar to the one of Digi 00x family. It's better to exclude the model from modalias of ALSA firewire-digi00x driver. This commit changes device entries so that the model is excluded. $ python3 crpp < ~/git/am-config-rom/misc/avid-adrenaline.img ROM header and bus information block ----------------------------------------------------------------- 400 04203a9c bus_info_length 4, crc_length 32, crc 15004 404 31333934 bus_name "1394" 408 e064a002 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 10 (2048) 40c 00a07e01 company_id 00a07e | 410 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 root directory ----------------------------------------------------------------- 414 0005d08c directory_length 5, crc 53388 418 0300a07e vendor 41c 8100000c --> descriptor leaf at 44c 420 0c008380 node capabilities 424 8d000002 --> eui-64 leaf at 42c 428 d1000004 --> unit directory at 438 eui-64 leaf at 42c ----------------------------------------------------------------- 42c 0002410f leaf_length 2, crc 16655 430 00a07e01 company_id 00a07e | 434 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 unit directory at 438 ----------------------------------------------------------------- 438 0004d6c9 directory_length 4, crc 54985 43c 1200a02d specifier id: 1394 TA 440 13014001 version: Vender Unique and AV/C 444 17000001 model 448 81000009 --> descriptor leaf at 46c descriptor leaf at 44c ----------------------------------------------------------------- 44c 00077205 leaf_length 7, crc 29189 450 00000000 textual descriptor 454 00000000 minimal ASCII 458 41766964 "Avid" 45c 20546563 " Tec" 460 686e6f6c "hnol" 464 6f677900 "ogy" 468 00000000 descriptor leaf at 46c ----------------------------------------------------------------- 46c 000599a5 leaf_length 5, crc 39333 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 41647265 "Adre" 47c 6e616c69 "nali" 480 6e650000 "ne" Reported-by: Simon Wood Fixes: 9edf723fd858 ("ALSA: firewire-digi00x: add skeleton for Digi 002/003 family") Cc: # 4.4+ Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200823075545.56305-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/digi00x/digi00x.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index c84b913a9fe0..ab8408966ec3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2"); #define VENDOR_DIGIDESIGN 0x00a07e #define MODEL_CONSOLE 0x000001 #define MODEL_RACK 0x000002 +#define SPEC_VERSION 0x000001 static int name_card(struct snd_dg00x *dg00x) { @@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = { /* Both of 002/003 use the same ID. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_CONSOLE, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_RACK, }, {} -- cgit v1.2.1 From 1965c4364bdd025e9e3142592edd76addcc4bab4 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Sun, 23 Aug 2020 18:58:50 +0800 Subject: ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620 If USB autosuspend is enabled, both front and rear panel can no longer detect jack insertion. Enable USB remote wakeup, i.e. needs_remote_wakeup = 1, doesn't help either. So disable USB autosuspend to prevent missing jack detection event. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20200823105854.26950-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 18 ++++++++++++++---- sound/usb/quirks.c | 10 ++++++++++ sound/usb/usbaudio.h | 1 + 3 files changed, 25 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..416de71c6895 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ { USB_DEVICE(0x17aa, 0x1046), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear", - "Lenovo-ThinkStation-P620-Rear"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Rear", + .profile_name = "Lenovo-ThinkStation-P620-Rear", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ { USB_DEVICE(0x17aa, 0x104d), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main", - "Lenovo-ThinkStation-P620-Main"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Main", + .profile_name = "Lenovo-ThinkStation-P620-Main", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Native Instruments MK2 series */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..b800fd92106c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, return 1; /* Continue with creating streams and mixer */ } +static int setup_disable_autosuspend(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + driver->supports_autosuspend = 0; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, + [QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend, }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b91c4c0807ec..6839915a0128 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -102,6 +102,7 @@ enum quirk_type { QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, QUIRK_SETUP_FMT_AFTER_RESUME, + QUIRK_SETUP_DISABLE_AUTOSUSPEND, QUIRK_TYPE_COUNT }; -- cgit v1.2.1 From ee0761d1d8222bcc5c86bf10849dc86cf008557c Mon Sep 17 00:00:00 2001 From: Tong Zhang Date: Mon, 24 Aug 2020 18:45:41 -0400 Subject: ALSA: ca0106: fix error code handling snd_ca0106_spi_write() returns 1 on error, snd_ca0106_pcm_power_dac() is returning the error code directly, and the caller is expecting an negative error code Signed-off-by: Tong Zhang Cc: Link: https://lore.kernel.org/r/20200824224541.1260307-1-ztong0001@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 70d775ff967e..c189f70c82cb 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, else /* Power down */ chip->spi_dac_reg[reg] |= bit; - return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; } return 0; } -- cgit v1.2.1 From 216116eae43963c662eb84729507bad95214ca6b Mon Sep 17 00:00:00 2001 From: Mohan Kumar Date: Tue, 25 Aug 2020 10:54:14 +0530 Subject: ALSA: hda: Fix 2 channel swapping for Tegra The Tegra HDA codec HW implementation has an issue related to not swapping the 2 channel Audio Sample Packet(ASP) channel mapping. Whatever the FL and FR mapping specified the left channel always comes out of left speaker and right channel on right speaker. So add condition to disallow the swapping of FL,FR during the playback. Signed-off-by: Mohan Kumar Acked-by: Sameer Pujar Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b8c8490e568b..3259d713ace9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3734,6 +3734,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3741,6 +3742,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } -- cgit v1.2.1 From 23d63a31d9f44d7daeac0d1fb65c6a73c70e5216 Mon Sep 17 00:00:00 2001 From: Mohan Kumar Date: Tue, 25 Aug 2020 10:54:15 +0530 Subject: ALSA: hda/tegra: Program WAKEEN register for Tegra The WAKEEN bits are used to indicate which bits in the STATESTS register may cause wake event during the codec state change request. Configure the WAKEEN register for the Tegra to detect the wake events. Signed-off-by: Mohan Kumar Acked-by: Sameer Pujar Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_tegra.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c94553bcca88..70164d1428d4 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); if (chip && chip->running) { + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); azx_enter_link_reset(chip); } @@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) if (chip && chip->running) { hda_tegra_init(hda); azx_init_chip(chip, 1); + /* disable controller wake up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); } return 0; -- cgit v1.2.1 From eed8f88b109aa927fbf0d0c80ff9f8d00444ca7f Mon Sep 17 00:00:00 2001 From: Tiezhu Yang Date: Tue, 25 Aug 2020 17:39:48 +0800 Subject: Revert "ALSA: hda: Add support for Loongson 7A1000 controller" This reverts commit 61eee4a7fc40 ("ALSA: hda: Add support for Loongson 7A1000 controller") to fix the following error on the Loongson LS7A platform: rcu: INFO: rcu_preempt self-detected stall on CPU NMI backtrace for cpu 0 CPU: 0 PID: 68 Comm: kworker/0:2 Not tainted 5.8.0+ #3 Hardware name: , BIOS Workqueue: events azx_probe_work [snd_hda_intel] Call Trace: [] show_stack+0x9c/0x130 [] dump_stack+0xb0/0xf0 [] nmi_cpu_backtrace+0x134/0x140 [] nmi_trigger_cpumask_backtrace+0x190/0x200 [] rcu_dump_cpu_stacks+0x12c/0x190 [] rcu_sched_clock_irq+0xa2c/0xfc8 [] update_process_times+0x2c/0xb8 [] tick_sched_timer+0x40/0xb8 [] __hrtimer_run_queues+0x118/0x1d0 [] hrtimer_interrupt+0x12c/0x2d8 [] c0_compare_interrupt+0x74/0xa0 [] __handle_irq_event_percpu+0xa8/0x198 [] handle_irq_event_percpu+0x30/0x90 [] handle_percpu_irq+0x88/0xb8 [] generic_handle_irq+0x44/0x60 [] do_IRQ+0x18/0x28 [] plat_irq_dispatch+0x64/0x100 [] handle_int+0x140/0x14c [] irq_exit+0xf8/0x100 Because AZX_DRIVER_GENERIC can not work well for Loongson LS7A HDA controller, it needs some workarounds which are not merged into the upstream kernel at this time, so it should revert this patch now. Fixes: 61eee4a7fc40 ("ALSA: hda: Add support for Loongson 7A1000 controller") Cc: # 5.9-rc1+ Signed-off-by: Tiezhu Yang Link: https://lore.kernel.org/r/1598348388-2518-1-git-send-email-yangtiezhu@loongson.cn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e34a4d5d047c..0f86e3765bb3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2745,8 +2745,6 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, - /* Loongson */ - { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- cgit v1.2.1 From 7c5b892e0871655fea3294ffac6fa3cc3400b60d Mon Sep 17 00:00:00 2001 From: Joshua Sivec Date: Tue, 25 Aug 2020 18:55:18 +0200 Subject: ALSA: usb-audio: Add implicit feedback quirk for UR22C This uses the same quirk as the Motu and SSL2 devices. Tested on the UR22C. Fixes bug 208851. Signed-off-by: Joshua Sivec BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208851 Cc: Link: https://lore.kernel.org/r/20200825165515.8239-1-sivec@posteo.net Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..10d342270dd9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -369,6 +369,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ + case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */ ep = 0x81; ifnum = 2; -- cgit v1.2.1 From 14335d8b9e1a2bf006f9d969a103f9731cabb210 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Franti=C5=A1ek=20Ku=C4=8Dera?= Date: Tue, 25 Aug 2020 17:31:13 +0200 Subject: ALSA: usb-audio: Add basic capture support for Pioneer DJ DJM-250MK2 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch extends support for DJM-250MK2 and allows recording. However, DVS is not possible yet (see the comment in code). Signed-off-by: František Kučera Cc: Link: https://lore.kernel.org/r/20200825153113.6352-1-konference@frantovo.cz Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 60 ++++++++++++++++++++++++++++++++++++++++++------ sound/usb/quirks.c | 1 + 3 files changed, 55 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 10d342270dd9..b401ee894e1b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -375,6 +375,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 416de71c6895..23eafd50126f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3559,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX + * + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3590,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .rate_max = 48000, .nr_rates = 1, .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b800fd92106c..75bbdc691243 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1503,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ -- cgit v1.2.1 From 8bcea6cb2cbc1f749e574954569323dec5e2920e Mon Sep 17 00:00:00 2001 From: Adrien Crivelli Date: Wed, 26 Aug 2020 17:40:14 +0900 Subject: ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion NT950XCJ-X716A The Galaxy Book Ion NT950XCJ-X716A (15 inches) uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug. I confirmed on my own hardware that this fixes the bug. This also correct the model name for the 13 inches version. It was incorrectly referenced as NT950XCJ-X716A in commit e17f02d05. But it should have been NP930XCJ-K01US. Fixes: e17f02d0559c ("ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Adrien Crivelli Cc: Link: https://lore.kernel.org/r/20200826084014.211217-1-adrien.crivelli@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..98789691a479 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7695,7 +7695,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- cgit v1.2.1 From 858e0ad9301d1270c02b5aca97537d2d6ee9dd68 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 26 Aug 2020 20:03:06 +0300 Subject: ALSA: hda/hdmi: always check pin power status in i915 pin fixup When system is suspended with active audio playback to HDMI/DP, two alternative sequences can happen at resume: a) monitor is detected first and ALSA prepare follows normal stream setup sequence, or b) ALSA prepare is called first, but monitor is not yet detected, so PCM is restarted without a pin, In case of (b), on i915 systems, haswell_verify_D0() is not called at resume and the pin power state may be incorrect. Result is lack of audio after resume with no error reported back to user-space. Fix the problem by always verifying converter and pin state in the i915_pin_cvt_fixup(). BugLink: https://github.com/thesofproject/linux/issues/2388 Signed-off-by: Kai Vehmanen Cc: Link: https://lore.kernel.org/r/20200826170306.701566-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3259d713ace9..1e1b13eb7829 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); -- cgit v1.2.1 From 15cbff3fbbc631952c346744f862fb294504b5e2 Mon Sep 17 00:00:00 2001 From: Dan Crawford Date: Sat, 29 Aug 2020 12:49:46 +1000 Subject: ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO Following Christian Lachner's patch for Gigabyte X570-based motherboards, also patch the MSI X570-A PRO motherboard; the ALC1220 codec requires the same workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. I suspect most if all X570 motherboards will require similar patches. [ The entries reordered in the SSID order -- tiwai ] Related buglink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Dan Crawford Cc: Link: https://lore.kernel.org/r/20200829024946.5691-1-dnlcrwfrd@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98789691a479..2ef8b080d84b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2475,6 +2475,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- cgit v1.2.1 From 949a1ebe8cea7b342085cb6a4946b498306b9493 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Sep 2020 15:18:02 +0200 Subject: ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check The PCM OSS mulaw plugin has a check of the format of the counter part whether it's a linear format. The check is with snd_BUG_ON() that emits WARN_ON() when the debug config is set, and it confuses syzkaller as if it were a serious issue. Let's drop snd_BUG_ON() for avoiding that. While we're at it, correct the error code to a more suitable, EINVAL. Reported-by: syzbot+23b22dc2e0b81cbfcc95@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/20200901131802.18157-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/mulaw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, -- cgit v1.2.1 From d1a58aae6648031afcb2a93f5af8498037efe2cc Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:12 +0530 Subject: ALSA: core: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-2-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/core/timer.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index d9f85f2d66a3..6e27d87b18ed 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -816,9 +816,9 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer, * timer tasklet * */ -static void snd_timer_tasklet(unsigned long arg) +static void snd_timer_tasklet(struct tasklet_struct *t) { - struct snd_timer *timer = (struct snd_timer *) arg; + struct snd_timer *timer = from_tasklet(timer, t, task_queue); unsigned long flags; if (timer->card && timer->card->shutdown) { @@ -967,8 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, INIT_LIST_HEAD(&timer->ack_list_head); INIT_LIST_HEAD(&timer->sack_list_head); spin_lock_init(&timer->lock); - tasklet_init(&timer->task_queue, snd_timer_tasklet, - (unsigned long)timer); + tasklet_setup(&timer->task_queue, snd_timer_tasklet); timer->max_instances = 1000; /* default limit per timer */ if (card != NULL) { timer->module = card->module; -- cgit v1.2.1 From 51bc581a21c420f71b4d78e5718efac2aaf854b1 Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:13 +0530 Subject: ALSA: firewire: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Acked-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20200902040221.354941-3-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8586f75441d..ee1c428b1fd3 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -64,7 +64,7 @@ #define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header. #define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing. -static void pcm_period_tasklet(unsigned long data); +static void pcm_period_tasklet(struct tasklet_struct *t); /** * amdtp_stream_init - initialize an AMDTP stream structure @@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); - tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s); + tasklet_setup(&s->period_tasklet, pcm_period_tasklet); s->packet_index = 0; init_waitqueue_head(&s->callback_wait); @@ -441,9 +441,9 @@ static void update_pcm_pointers(struct amdtp_stream *s, } } -static void pcm_period_tasklet(unsigned long data) +static void pcm_period_tasklet(struct tasklet_struct *t) { - struct amdtp_stream *s = (void *)data; + struct amdtp_stream *s = from_tasklet(s, t, period_tasklet); struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); if (pcm) -- cgit v1.2.1 From 70f8b2f12dc033b24e2932c2b196d1ca4a915417 Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:14 +0530 Subject: ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-4-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 023c35a2a951..35e76480306e 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -921,10 +921,10 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) add_timer(&dpcm->timer); } -static void snd_card_asihpi_int_task(unsigned long data) +static void snd_card_asihpi_int_task(struct tasklet_struct *t) { - struct hpi_adapter *a = (struct hpi_adapter *)data; - struct snd_card_asihpi *asihpi; + struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t); + struct hpi_adapter *a = asihpi->hpi; WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; @@ -2871,8 +2871,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, if (hpi->interrupt_mode) { asihpi->pcm_start = snd_card_asihpi_pcm_int_start; asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop; - tasklet_init(&asihpi->t, snd_card_asihpi_int_task, - (unsigned long)hpi); + tasklet_setup(&asihpi->t, snd_card_asihpi_int_task); hpi->interrupt_callback = snd_card_asihpi_isr; } else { asihpi->pcm_start = snd_card_asihpi_pcm_timer_start; -- cgit v1.2.1 From c2082393d55487d30d1b307242135b2b442920a0 Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:15 +0530 Subject: ALSA: riptide: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-5-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b4f300281822..098c69b3b7aa 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1070,9 +1070,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, return 0; } -static void riptide_handleirq(unsigned long dev_id) +static void riptide_handleirq(struct tasklet_struct *t) { - struct snd_riptide *chip = (void *)dev_id; + struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq); struct cmdif *cif = chip->cif; struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1]; struct snd_pcm_runtime *runtime; @@ -1843,7 +1843,7 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci, chip->received_irqs = 0; chip->handled_irqs = 0; chip->cif = NULL; - tasklet_init(&chip->riptide_tq, riptide_handleirq, (unsigned long)chip); + tasklet_setup(&chip->riptide_tq, riptide_handleirq); if ((chip->res_port = request_region(chip->port, 64, "RIPTIDE")) == NULL) { -- cgit v1.2.1 From 1a1575a151478f336c473137c32b82a3933e402e Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:16 +0530 Subject: ALSA: hdsp: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-6-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 6 +++--- sound/pci/rme9652/hdspm.c | 7 +++---- 2 files changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 227aece17e39..dda56ecfd33b 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3791,9 +3791,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) return 0; } -static void hdsp_midi_tasklet(unsigned long arg) +static void hdsp_midi_tasklet(struct tasklet_struct *t) { - struct hdsp *hdsp = (struct hdsp *)arg; + struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet); if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); @@ -5182,7 +5182,7 @@ static int snd_hdsp_create(struct snd_card *card, spin_lock_init(&hdsp->lock); - tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); + tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet); pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0fa49f4d15cf..572350aaf18d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card, } -static void hdspm_midi_tasklet(unsigned long arg) +static void hdspm_midi_tasklet(struct tasklet_struct *t) { - struct hdspm *hdspm = (struct hdspm *)arg; + struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet); int i = 0; while (i < hdspm->midiPorts) { @@ -6836,8 +6836,7 @@ static int snd_hdspm_create(struct snd_card *card, } - tasklet_init(&hdspm->midi_tasklet, - hdspm_midi_tasklet, (unsigned long) hdspm); + tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet); if (hdspm->io_type != MADIface) { -- cgit v1.2.1 From b07e4a3fb2473e3e55e209e2cd22295276de333a Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:17 +0530 Subject: ASoC: fsl_esai: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Acked-by: Mark Brown Link: https://lore.kernel.org/r/20200902040221.354941-7-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_esai.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 4ae36099ae82..79b861afd986 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -708,9 +708,9 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx) ESAI_xFCR_xFR, 0); } -static void fsl_esai_hw_reset(unsigned long arg) +static void fsl_esai_hw_reset(struct tasklet_struct *t) { - struct fsl_esai *esai_priv = (struct fsl_esai *)arg; + struct fsl_esai *esai_priv = from_tasklet(esai_priv, t, task); bool tx = true, rx = false, enabled[2]; unsigned long lock_flags; u32 tfcr, rfcr; @@ -1070,8 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - tasklet_init(&esai_priv->task, fsl_esai_hw_reset, - (unsigned long)esai_priv); + tasklet_setup(&esai_priv->task, fsl_esai_hw_reset); pm_runtime_enable(&pdev->dev); -- cgit v1.2.1 From ea0364c01e9539a2ec973ab49df4b3124ec5d0fc Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:18 +0530 Subject: ASoC: siu: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Acked-by: Mark Brown Link: https://lore.kernel.org/r/20200902040221.354941-8-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/soc/sh/siu_pcm.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index bd9de77c35f3..50fc7810723e 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -198,9 +198,9 @@ static int siu_pcm_rd_set(struct siu_port *port_info, return 0; } -static void siu_io_tasklet(unsigned long data) +static void siu_io_tasklet(struct tasklet_struct *t) { - struct siu_stream *siu_stream = (struct siu_stream *)data; + struct siu_stream *siu_stream = from_tasklet(siu_stream, t, tasklet); struct snd_pcm_substream *substream = siu_stream->substream; struct device *dev = substream->pcm->card->dev; struct snd_pcm_runtime *rt = substream->runtime; @@ -520,10 +520,8 @@ static int siu_pcm_new(struct snd_soc_component *component, (*port_info)->pcm = pcm; /* IO tasklets */ - tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet, - (unsigned long)&(*port_info)->playback); - tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet, - (unsigned long)&(*port_info)->capture); + tasklet_setup(&(*port_info)->playback.tasklet, siu_io_tasklet); + tasklet_setup(&(*port_info)->capture.tasklet, siu_io_tasklet); } dev_info(card->dev, "SuperH SIU driver initialized.\n"); -- cgit v1.2.1 From ad77b35abb53481bc4cd7237ceae14af7275a50d Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:19 +0530 Subject: ASoC: txx9: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Acked-by: Mark Brown Link: https://lore.kernel.org/r/20200902040221.354941-9-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/soc/txx9/txx9aclc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 4b1cd4da3e36..939b33ec39f5 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -134,9 +134,9 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) #define NR_DMA_CHAIN 2 -static void txx9aclc_dma_tasklet(unsigned long data) +static void txx9aclc_dma_tasklet(struct tasklet_struct *t) { - struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct txx9aclc_dmadata *dmadata = from_tasklet(dmadata, t, tasklet); struct dma_chan *chan = dmadata->dma_chan; struct dma_async_tx_descriptor *desc; struct snd_pcm_substream *substream = dmadata->substream; @@ -352,8 +352,7 @@ static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, "playback" : "capture"); return -EBUSY; } - tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, - (unsigned long)dmadata); + tasklet_setup(&dmadata->tasklet, txx9aclc_dma_tasklet); return 0; } -- cgit v1.2.1 From 7fbf9547f3420bae7f3713fc97349767bced5906 Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:20 +0530 Subject: ALSA: usb-audio: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-10-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index df639fe03118..e8287a05e36b 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -344,10 +344,9 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep) spin_unlock_irqrestore(&ep->buffer_lock, flags); } -static void snd_usbmidi_out_tasklet(unsigned long data) +static void snd_usbmidi_out_tasklet(struct tasklet_struct *t) { - struct snd_usb_midi_out_endpoint *ep = - (struct snd_usb_midi_out_endpoint *) data; + struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet); snd_usbmidi_do_output(ep); } @@ -1441,7 +1440,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, } spin_lock_init(&ep->buffer_lock); - tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep); + tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet); init_waitqueue_head(&ep->drain_wait); for (i = 0; i < 0x10; ++i) -- cgit v1.2.1 From 07da90b0e5fc299c0ce4d34d1916d7a79b86848b Mon Sep 17 00:00:00 2001 From: Allen Pais Date: Wed, 2 Sep 2020 09:32:21 +0530 Subject: ALSA: ua101: convert tasklets to use new tasklet_setup() API In preparation for unconditionally passing the struct tasklet_struct pointer to all tasklet callbacks, switch to using the new tasklet_setup() and from_tasklet() to pass the tasklet pointer explicitly. Signed-off-by: Romain Perier Signed-off-by: Allen Pais Link: https://lore.kernel.org/r/20200902040221.354941-11-allen.lkml@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 884e740a785c..3b2dce1043f5 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua, *value -= ua->playback.queue_length; } -static void playback_tasklet(unsigned long data) +static void playback_tasklet(struct tasklet_struct *t) { - struct ua101 *ua = (void *)data; + struct ua101 *ua = from_tasklet(ua, t, playback_tasklet); unsigned long flags; unsigned int frames; struct ua101_urb *urb; @@ -1218,8 +1218,7 @@ static int ua101_probe(struct usb_interface *interface, spin_lock_init(&ua->lock); mutex_init(&ua->mutex); INIT_LIST_HEAD(&ua->ready_playback_urbs); - tasklet_init(&ua->playback_tasklet, - playback_tasklet, (unsigned long)ua); + tasklet_setup(&ua->playback_tasklet, playback_tasklet); init_waitqueue_head(&ua->alsa_capture_wait); init_waitqueue_head(&ua->rate_feedback_wait); init_waitqueue_head(&ua->alsa_playback_wait); -- cgit v1.2.1 From f804a324a41a880c1ab43cc5145d8b3e5790430d Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 2 Sep 2020 18:42:07 +0300 Subject: ALSA: hda: hdmi - add Rocketlake support Add Rocketlake HDMI codec support. Rocketlake shares the pin-to-port mapping table with Tigerlake. Signed-off-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200902154207.1440393-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1e1b13eb7829..402050088090 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- cgit v1.2.1 From 13774d81f38538c5fa2924bdcdfa509155480fa6 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Wed, 2 Sep 2020 18:42:18 +0300 Subject: ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled In snd_hdac_device_init pm_runtime_set_active is called to increase child_count in parent device. But when it is failed to build connection with GPU for one case that integrated graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be invoked to clean up a HD-audio extended codec base device. At this time the child_count of parent is not decreased, which makes parent device can't get suspended. This patch calls pm_runtime_set_suspended to decrease child_count in parent device in snd_hdac_device_exit to match with snd_hdac_device_init. pm_runtime_set_suspended can make sure that it will not decrease child_count if the device is already suspended. Signed-off-by: Rander Wang Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Guennadi Liakhovetski Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200902154218.1440441-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..3e9e9ac804f6 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); -- cgit v1.2.1 From ae035947162c9350619de1b3a3e3051a265f43f2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 2 Sep 2020 18:42:39 +0300 Subject: ALSA: hda: add dev_dbg log when driver is not selected On SKL+ Intel platforms, the driver selection is handled by the snd_intel_dspcfg, and when the HDaudio legacy driver is not selected, be it with the auto-selection or user preferences with a kernel parameter, the probe aborts with no logs, only a -ENODEV return value. Having no dmesg trace, even with dynamic debug enabled, makes support more complicated than it needs to be, and even experienced users can be fooled. A simple dev_dbg() trace solves this problem. BugLink: https://github.com/thesofproject/linux/issues/2330 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Guennadi Liakhovetski Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200902154239.1440537-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0f86e3765bb3..36a9dbc33aa0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci, */ if (dmic_detect) { err = snd_intel_dsp_driver_probe(pci); - if (err != SND_INTEL_DSP_DRIVER_ANY && - err != SND_INTEL_DSP_DRIVER_LEGACY) + if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) { + dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n"); return -ENODEV; + } } else { dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n"); } -- cgit v1.2.1 From b79de57b4378a93115307be6962d05b099eb0f37 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 2 Sep 2020 18:42:50 +0300 Subject: ALSA: hda: use consistent HDAudio spelling in comments/docs We use HDaudio and HDAudio, pick one to make searches easier. No functionality change Also fix timestamping typo in documentation. Reported-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Guennadi Liakhovetski Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200902154250.1440585-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 10 +++++----- sound/x86/Kconfig | 2 +- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 99aec7349167..1c5114dedda9 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -54,7 +54,7 @@ static const struct config_entry config_table[] = { #endif /* * Apollolake (Broxton-P) - * the legacy HDaudio driver is used except on Up Squared (SOF) and + * the legacy HDAudio driver is used except on Up Squared (SOF) and * Chromebooks (SST) */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -89,7 +89,7 @@ static const struct config_entry config_table[] = { }, #endif /* - * Skylake and Kabylake use legacy HDaudio driver except for Google + * Skylake and Kabylake use legacy HDAudio driver except for Google * Chromebooks (SST) */ @@ -135,7 +135,7 @@ static const struct config_entry config_table[] = { #endif /* - * Geminilake uses legacy HDaudio driver except for Google + * Geminilake uses legacy HDAudio driver except for Google * Chromebooks */ /* Geminilake */ @@ -157,7 +157,7 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDaudio driver except for Google Chromebooks and when DMICs are + * HDAudio driver except for Google Chromebooks and when DMICs are * present. Two cases are required since Coreboot does not expose NHLT * tables. * @@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (pci->class == 0x040300) return SND_INTEL_DSP_DRIVER_LEGACY; if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class); return SND_INTEL_DSP_DRIVER_LEGACY; } diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 77777192f650..4ffcc5e623c2 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,7 +9,7 @@ menuconfig SND_X86 if SND_X86 config HDMI_LPE_AUDIO - tristate "HDMI audio without HDaudio on Intel Atom platforms" + tristate "HDMI audio without HDAudio on Intel Atom platforms" depends on DRM_I915 select SND_PCM help -- cgit v1.2.1 From 6a6660d049f88b89fd9a4b9db3581b245f7782fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 3 Sep 2020 10:33:00 +0200 Subject: ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen There've been quite a few regression reports about the lowered volume (reduced to ca 65% from the previous level) on Lenovo Thinkpad X1 after the commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Although the commit itself does the right thing from HD-audio POV in order to have a volume control for bass speakers, it seems that the machine has some secret recipe under the hood. Through experiments, Benjamin Poirier found out that the following routing gives the best result: * DAC1 (NID 0x02) -> Speaker pin (NID 0x14) * DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) & Headphone pin (0x21) * DAC3 (NID 0x06) -> Unused DAC1 seems to have some equalizer internally applied, and you'd get again the output in a bad quality if you connect this to the headphone pin. Hence the headphone is connected to DAC2, which is now shared with the bass speaker pin. DAC3 has no volume amp, hence it's not connected at all. For achieving the routing above, this patch introduced a couple of workarounds: * The connection list of bass speaker pin (NID 0x17) is reduced not to include DAC3 (NID 0x06) * Pass preferred_pairs array to specify the fixed connection Here, both workarounds are needed because the generic parser prefers the individual DAC assignment over others. When the routing above is applied, the generic parser creates the two volume controls "Front" and "Bass Speaker". Since we have only two DACs for three output pins, those are not fully controlling each output individually, and it would confuse PulseAudio. For avoiding the pitfall, in this patch, we rename those volume controls to some unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and concentrate only on the still good-working "Master" volume control. If a user still wants to control each DAC volume, they can still change manually via "DAC1" and "DAC2" volume controls. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Reported-by: Benjamin Poirier Reviewed-by: Jaroslav Kysela Tested-by: Benjamin Poirier Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3214171 BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3276276 Link: https://lore/kernel.org/r/20200829112746.3118-1-benjamin.poirier@gmail.com Link: https://lore.kernel.org/r/20200903083300.6333-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ef8b080d84b..c521a1f17096 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5868,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec, } } +/* Quirk for Thinkpad X1 7th and 8th Gen + * The following fixed routing needed + * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly + * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC + * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp + */ +static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + case HDA_FIXUP_ACT_BUILD: + /* The generic parser creates somewhat unintuitive volume ctls + * with the fixed routing above, and the shared DAC2 may be + * confusing for PA. + * Rename those to unique names so that PA doesn't touch them + * and use only Master volume. + */ + rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume"); + rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume"); + break; + } +} + static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -6136,6 +6169,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, + ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, @@ -7281,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_X1_GEN7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_thinkpad_x1_gen7, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 + .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7 }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, -- cgit v1.2.1