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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 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/pci') 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 v1.2.3 From c3cdf189276c2a63da62ee250615bd55e3fb680d Mon Sep 17 00:00:00 2001 From: Luke D Jones Date: Mon, 7 Sep 2020 20:19:59 +1200 Subject: ALSA: hda: fixup headset for ASUS GX502 laptop The GX502 requires a few steps to enable the headset i/o: pincfg, verbs to enable and unmute the amp used for headpone out, and a jacksense callback to toggle output via internal or jack using a verb. Signed-off-by: Luke D Jones Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208005 Link: https://lore.kernel.org/r/20200907081959.56186-1-luke@ljones.dev Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 65 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c521a1f17096..abfc602c3b92 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5993,6 +5993,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) + alc_write_coef_idx(codec, 0x10, 0x8a20); + else + alc_write_coef_idx(codec, 0x10, 0x0a20); +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6173,6 +6207,9 @@ enum { ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, @@ -7338,6 +7375,33 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* front HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7711,6 +7775,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.2.3 From fc19d559b0d31b5b831fd468b10d7dadafc0d0ec Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 9 Sep 2020 10:00:41 +0800 Subject: ALSA: hda/realtek - The Mic on a RedmiBook doesn't work The Mic connects to the Nid 0x19, but the configuration of Nid 0x19 is not defined to Mic, and also need to set the coeff to enable the auto detection on the Nid 0x19. After this change, the Mic plugging in or plugging out could be detected and could record the sound from the Mic. And the coeff value is suggested by Kailang of Realtek. Cc: Kailang Yang Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20200909020041.8967-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index abfc602c3b92..85e207173f5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6228,6 +6228,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC255_FIXUP_XIAOMI_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7591,6 +7592,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { } + }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7888,6 +7899,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), @@ -8065,6 +8077,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- cgit v1.2.3