From a7663c89f4193dbf717572e46e5a3251940dbdc8 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Sat, 19 Mar 2022 20:03:25 +0800 Subject: ASoC: mediatek: mt6358: add missing EXPORT_SYMBOLs Fixes the following build errors when mt6358 is configured as module: >> ERROR: modpost: "mt6358_set_mtkaif_protocol" >> [sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.ko] undefined! >> ERROR: modpost: "mt6358_set_mtkaif_protocol" >> [sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.ko] undefined! Fixes: 6a8d4198ca80 ("ASoC: mediatek: mt6358: add codec driver") Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220319120325.11882-1-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6358.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 9b263a9a669d..4c7b5d940799 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -107,6 +107,7 @@ int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt, priv->mtkaif_protocol = mtkaif_protocol; return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_protocol); static void playback_gpio_set(struct mt6358_priv *priv) { @@ -273,6 +274,7 @@ int mt6358_mtkaif_calibration_enable(struct snd_soc_component *cmpnt) 1 << RG_AUD_PAD_TOP_DAT_MISO_LOOPBACK_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_enable); int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) { @@ -296,6 +298,7 @@ int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) capture_gpio_reset(priv); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_disable); int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, int phase_1, int phase_2) @@ -310,6 +313,7 @@ int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, phase_2 << RG_AUD_PAD_TOP_PHASE_MODE2_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_calibration_phase); /* dl pga gain */ enum { -- cgit v1.2.3-58-ga151 From 5cb90dcb6ad569f4968da6dd841db10b91df5642 Mon Sep 17 00:00:00 2001 From: Meng Tang Date: Mon, 21 Mar 2022 14:57:54 +0800 Subject: ASoC: fsl-asoc-card: Fix jack_event() always return 0 Today, hp_jack_event and mic_jack_event always return 0. However, snd_soc_dapm_disable_pin and snd_soc_dapm_enable_pin may return a non-zero value, this will cause the user who calling hp_jack_event and mic_jack_event don't know whether the operation was really successfully. Signed-off-by: Meng Tang Acked-by: Shengjiu Wang Reviewed-by: Christophe Leroy Link: https://lore.kernel.org/r/20220321065754.18307-1-tangmeng@uniontech.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 370bc790c6ba..d9a0d4768c4d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -462,11 +462,9 @@ static int hp_jack_event(struct notifier_block *nb, unsigned long event, if (event & SND_JACK_HEADPHONE) /* Disable speaker if headphone is plugged in */ - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); + return snd_soc_dapm_disable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - - return 0; + return snd_soc_dapm_enable_pin(dapm, "Ext Spk"); } static struct notifier_block hp_jack_nb = { @@ -481,11 +479,9 @@ static int mic_jack_event(struct notifier_block *nb, unsigned long event, if (event & SND_JACK_MICROPHONE) /* Disable dmic if microphone is plugged in */ - snd_soc_dapm_disable_pin(dapm, "DMIC"); + return snd_soc_dapm_disable_pin(dapm, "DMIC"); else - snd_soc_dapm_enable_pin(dapm, "DMIC"); - - return 0; + return snd_soc_dapm_enable_pin(dapm, "DMIC"); } static struct notifier_block mic_jack_nb = { -- cgit v1.2.3-58-ga151 From 2f45a4e2897793cc6ae25f5fe78b485ce7fd01d0 Mon Sep 17 00:00:00 2001 From: Meng Tang Date: Fri, 18 Mar 2022 18:01:46 +0800 Subject: ASoC: rockchip: i2s_tdm: Fixup config for SND_SOC_DAIFMT_DSP_A/B SND_SOC_DAIFMT_DSP_A: PCM delay 1 bit mode, L data MSB after FRM LRC SND_SOC_DAIFMT_DSP_B: PCM no delay mode, L data MSB during FRM LRC Fixes: 081068fd64140 (ASoC: rockchip: add support for i2s-tdm controller) Signed-off-by: Meng Tang Link: https://lore.kernel.org/r/20220318100146.23991-1-tangmeng@uniontech.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s_tdm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index d3b710406941..98700e75b82a 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -469,14 +469,14 @@ static int rockchip_i2s_tdm_set_fmt(struct snd_soc_dai *cpu_dai, txcr_val = I2S_TXCR_IBM_NORMAL; rxcr_val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - txcr_val = I2S_TXCR_TFS_PCM; - rxcr_val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 mode */ txcr_val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); rxcr_val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + txcr_val = I2S_TXCR_TFS_PCM; + rxcr_val = I2S_RXCR_TFS_PCM; + break; default: ret = -EINVAL; goto err_pm_put; -- cgit v1.2.3-58-ga151 From ce18f905a500879e86ca998963a55f99d413a462 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Thu, 24 Mar 2022 14:21:58 +0800 Subject: ALSA: hda/realtek: Add mute and micmut LED support for Zbook Fury 17 G9 Zbook Fury 17 G9 requires the same ALC285_FIXUP_HP_GPIO_LED quirk to make its audio LEDs work. So apply the quirk, and make it the last one since it's an LED quirk. Fixes: 07bcab93946c ("ALSA: hda/realtek: Add support for HP Laptops") Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20220324062159.241313-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6ee67f41c45..e88fbef57c40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7011,6 +7011,7 @@ enum { ALC287_FIXUP_LEGION_16ACHG6, ALC287_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L41_SPI_2, + ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, @@ -8776,6 +8777,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_two, }, + [ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_spi_two, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC245_FIXUP_CS35L41_SPI_4] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_four, @@ -9031,7 +9038,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- cgit v1.2.3-58-ga151 From 664d66dc0a64b32e60a5ad59a9aebb08676a612b Mon Sep 17 00:00:00 2001 From: Zheng Bin Date: Wed, 23 Mar 2022 17:25:01 +0800 Subject: ASoC: SOF: Intel: Fix build error without SND_SOC_SOF_PCI_DEV If SND_SOC_SOF_PCI_DEV is n, bulding fails: sound/soc/sof/intel/pci-tng.o:(.data+0x1c0): undefined reference to `sof_pci_probe' sound/soc/sof/intel/pci-tng.o:(.data+0x1c8): undefined reference to `sof_pci_remove' sound/soc/sof/intel/pci-tng.o:(.data+0x1e0): undefined reference to `sof_pci_shutdown' sound/soc/sof/intel/pci-tng.o:(.data+0x290): undefined reference to `sof_pci_pm' Make SND_SOC_SOF_MERRIFIELD select SND_SOC_SOF_PCI_DEV to fix this. Fixes: 8d4ba1be3d22 ("ASoC: SOF: pci: split PCI into different drivers") Reported-by: Hulk Robot Signed-off-by: Zheng Bin Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220323092501.145879-1-zhengbin13@huawei.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index b53f216d4ecc..172419392b33 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -84,6 +84,7 @@ if SND_SOC_SOF_PCI config SND_SOC_SOF_MERRIFIELD tristate "SOF support for Tangier/Merrifield" default SND_SOC_SOF_PCI + select SND_SOC_SOF_PCI_DEV select SND_SOC_SOF_INTEL_ATOM_HIFI_EP help This adds support for Sound Open Firmware for Intel(R) platforms -- cgit v1.2.3-58-ga151 From 5a8738571747c1e275a40b69a608657603867b7e Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Sat, 26 Mar 2022 00:05:00 +0800 Subject: ALSA: hda/realtek: Enable headset mic on Lenovo P360 Lenovo P360 is another platform equipped with ALC897, and it needs ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20220325160501.705221-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e88fbef57c40..4c33cb57963d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11152,6 +11152,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), -- cgit v1.2.3-58-ga151 From 0112f822f8a6d8039c94e0bc9b264d7ffc5d4704 Mon Sep 17 00:00:00 2001 From: Xiaomeng Tong Date: Sun, 27 Mar 2022 14:08:22 +0800 Subject: ALSA: cs4236: fix an incorrect NULL check on list iterator The bug is here: err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); The list iterator value 'cdev' will *always* be set and non-NULL by list_for_each_entry(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found. To fix the bug, use a new variable 'iter' as the list iterator, while use the original variable 'cdev' as a dedicated pointer to point to the found element. And snd_card_cs423x_pnp() itself has NULL check for cdev. Cc: stable@vger.kernel.org Fixes: c2b73d1458014 ("ALSA: cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection") Signed-off-by: Xiaomeng Tong Link: https://lore.kernel.org/r/20220327060822.4735-1-xiam0nd.tong@gmail.com Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index b6bdebd9ef27..10112e1bb25d 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -494,7 +494,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static int dev; int err; struct snd_card *card; - struct pnp_dev *cdev; + struct pnp_dev *cdev, *iter; char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) @@ -510,9 +510,11 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, strcpy(cid, pdev->id[0].id); cid[5] = '1'; cdev = NULL; - list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { - if (!strcmp(cdev->id[0].id, cid)) + list_for_each_entry(iter, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(iter->id[0].id, cid)) { + cdev = iter; break; + } } err = snd_cs423x_card_new(&pdev->dev, dev, &card); if (err < 0) -- cgit v1.2.3-58-ga151 From 8a7724535bacbb94fd9441ec232a83d71006d2a9 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 28 Mar 2022 12:56:09 +0100 Subject: ALSA: hda/cs8409: Fix Warlock to use mono mic configuration Warlock/Bullseye Laptops have a mono DMIC, Cyborg uses a stereo DMIC, and the configuration should reflect this. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-2-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index aff2b5abb81e..1411e3845f16 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -907,8 +907,8 @@ static void cs8409_cs42l42_hw_init(struct hda_codec *codec) } /* DMIC1_MO=00b, DMIC1/2_SR=1 */ - if (codec->fixup_id == CS8409_WARLOCK || codec->fixup_id == CS8409_CYBORG) - cs8409_vendor_coef_set(codec, 0x09, 0x0003); + if (codec->fixup_id == CS8409_CYBORG) + cs8409_vendor_coef_set(codec, CS8409_DMIC_CFG, 0x0003); cs42l42_resume(cs42l42); -- cgit v1.2.3-58-ga151 From bdc159dfda0acec5ca3adde1a1b58e1e0ddc8311 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 28 Mar 2022 12:56:10 +0100 Subject: ALSA: hda/cs8409: Re-order quirk table into ascending order To ensure consistency, the quirk table should be re-ordered in ascending order [ a typo fix in the patch description by tiwai ] Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-3-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 2d1fa706327b..9c1fa97100ef 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -478,28 +478,29 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0A29, "Bullseye", CS8409_BULLSEYE), SND_PCI_QUIRK(0x1028, 0x0A2A, "Bullseye", CS8409_BULLSEYE), SND_PCI_QUIRK(0x1028, 0x0A2B, "Bullseye", CS8409_BULLSEYE), + SND_PCI_QUIRK(0x1028, 0x0A77, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A78, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A79, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A7A, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A7D, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A7E, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A7F, "Cyborg", CS8409_CYBORG), + SND_PCI_QUIRK(0x1028, 0x0A80, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AB0, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AB2, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AB1, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AB3, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AB4, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AB5, "Warlock", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0ACF, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0AD0, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0AD1, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0AD2, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0AD3, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0AD9, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0ADA, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0ADB, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0ADC, "Warlock", CS8409_WARLOCK), - SND_PCI_QUIRK(0x1028, 0x0AF4, "Warlock", CS8409_WARLOCK), - SND_PCI_QUIRK(0x1028, 0x0AF5, "Warlock", CS8409_WARLOCK), - SND_PCI_QUIRK(0x1028, 0x0BB5, "Warlock N3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), - SND_PCI_QUIRK(0x1028, 0x0BB6, "Warlock V3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), - SND_PCI_QUIRK(0x1028, 0x0A77, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A78, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A79, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A7A, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A7D, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A7E, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A7F, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0A80, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0ADF, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AE0, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AE1, "Cyborg", CS8409_CYBORG), @@ -512,11 +513,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0AEE, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AEF, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AF0, "Cyborg", CS8409_CYBORG), - SND_PCI_QUIRK(0x1028, 0x0AD0, "Dolphin", CS8409_DOLPHIN), - SND_PCI_QUIRK(0x1028, 0x0AD1, "Dolphin", CS8409_DOLPHIN), - SND_PCI_QUIRK(0x1028, 0x0AD2, "Dolphin", CS8409_DOLPHIN), - SND_PCI_QUIRK(0x1028, 0x0AD3, "Dolphin", CS8409_DOLPHIN), - SND_PCI_QUIRK(0x1028, 0x0ACF, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0AF4, "Warlock", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0AF5, "Warlock", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0BB5, "Warlock N3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0BB6, "Warlock V3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), {} /* terminator */ }; -- cgit v1.2.3-58-ga151 From 342b6b610ae2a351de904022271e740c4c2b452b Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 28 Mar 2022 12:56:11 +0100 Subject: ALSA: hda/cs8409: Fix Full Scale Volume setting for all variants All current variants (Bullseye/Warlock/Cyborg) should be using reduced volume (-6dB) for better speaker protection. Refactor to make more explicit the meaning and setting of Full Scale Volume setting to avoid future confusion. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-4-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 29 ++++++++++++++++------------- sound/pci/hda/patch_cs8409.h | 3 +++ 2 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 1411e3845f16..163ff3b3092a 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -733,6 +733,7 @@ static void cs42l42_resume(struct sub_codec *cs42l42) { 0x130A, 0x00 }, { 0x130F, 0x00 }, }; + int fsv_old, fsv_new; /* Bring CS42L42 out of Reset */ gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); @@ -749,8 +750,13 @@ static void cs42l42_resume(struct sub_codec *cs42l42) /* Clear interrupts, by reading interrupt status registers */ cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs)); - if (cs42l42->full_scale_vol) - cs8409_i2c_write(cs42l42, 0x2001, 0x01); + fsv_old = cs8409_i2c_read(cs42l42, 0x2001); + if (cs42l42->full_scale_vol == CS42L42_FULL_SCALE_VOL_0DB) + fsv_new = fsv_old & ~CS42L42_FULL_SCALE_VOL_MASK; + else + fsv_new = fsv_old & CS42L42_FULL_SCALE_VOL_MASK; + if (fsv_new != fsv_old) + cs8409_i2c_write(cs42l42, 0x2001, fsv_new); /* we have to explicitly allow unsol event handling even during the * resume phase so that the jack event is processed properly @@ -997,21 +1003,15 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, * Additionally set HSBIAS_SENSE_EN and Full Scale volume for some variants. */ switch (codec->fixup_id) { - case CS8409_WARLOCK: - spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; - spec->scodecs[CS8409_CODEC0]->full_scale_vol = 1; - break; - case CS8409_BULLSEYE: - spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; - spec->scodecs[CS8409_CODEC0]->full_scale_vol = 0; - break; case CS8409_CYBORG: spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x00a0; - spec->scodecs[CS8409_CODEC0]->full_scale_vol = 1; + spec->scodecs[CS8409_CODEC0]->full_scale_vol = + CS42L42_FULL_SCALE_VOL_MINUS6DB; break; default: - spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0003; - spec->scodecs[CS8409_CODEC0]->full_scale_vol = 1; + spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; + spec->scodecs[CS8409_CODEC0]->full_scale_vol = + CS42L42_FULL_SCALE_VOL_MINUS6DB; break; } @@ -1222,6 +1222,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID); + spec->scodecs[CS8409_CODEC0]->full_scale_vol = CS42L42_FULL_SCALE_VOL_MINUS6DB; + spec->scodecs[CS8409_CODEC1]->full_scale_vol = CS42L42_FULL_SCALE_VOL_MINUS6DB; + break; case HDA_FIXUP_ACT_PROBE: /* Fix Sample Rate to 48kHz */ diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index d0b725c7285b..8e846f292cd0 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -235,6 +235,9 @@ enum cs8409_coefficient_index_registers { #define CS42L42_I2C_SLEEP_US (2000) #define CS42L42_PDN_TIMEOUT_US (250000) #define CS42L42_PDN_SLEEP_US (2000) +#define CS42L42_FULL_SCALE_VOL_MASK (2) +#define CS42L42_FULL_SCALE_VOL_0DB (1) +#define CS42L42_FULL_SCALE_VOL_MINUS6DB (0) /* Dell BULLSEYE / WARLOCK / CYBORG Specific Definitions */ -- cgit v1.2.3-58-ga151 From 6581a045d54c6a8fe335dd2f343fc7cd2ebfe9e7 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 28 Mar 2022 12:56:12 +0100 Subject: ALSA: hda/cs8409: Support new Warlock MLK Variants Added 15 new laptops, with 2 variants: Warlock MLK and Warlock MLK with Dual Mic The only difference between the variants, is the the dual Mic variants use a stereo DMIC. These variants do no use reduce volume (Full Scale Volume) Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-5-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 29 +++++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.c | 15 +++++++++++++-- sound/pci/hda/patch_cs8409.h | 2 ++ 3 files changed, 44 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 9c1fa97100ef..8d20d7fb3d68 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -515,8 +515,23 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0AF0, "Cyborg", CS8409_CYBORG), SND_PCI_QUIRK(0x1028, 0x0AF4, "Warlock", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0AF5, "Warlock", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0B92, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0B93, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0B94, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0B95, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0B96, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0B97, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0BB2, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0BB3, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0BB4, "Warlock MLK", CS8409_WARLOCK_MLK), SND_PCI_QUIRK(0x1028, 0x0BB5, "Warlock N3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), SND_PCI_QUIRK(0x1028, 0x0BB6, "Warlock V3 15 TGL-U Nuvoton EC", CS8409_WARLOCK), + SND_PCI_QUIRK(0x1028, 0x0BB8, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0BB9, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0BBA, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0BBB, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0BBC, "Warlock MLK", CS8409_WARLOCK_MLK), + SND_PCI_QUIRK(0x1028, 0x0BBD, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), {} /* terminator */ }; @@ -524,6 +539,8 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { const struct hda_model_fixup cs8409_models[] = { { .id = CS8409_BULLSEYE, .name = "bullseye" }, { .id = CS8409_WARLOCK, .name = "warlock" }, + { .id = CS8409_WARLOCK_MLK, .name = "warlock mlk" }, + { .id = CS8409_WARLOCK_MLK_DUAL_MIC, .name = "warlock mlk dual mic" }, { .id = CS8409_CYBORG, .name = "cyborg" }, { .id = CS8409_DOLPHIN, .name = "dolphin" }, {} @@ -542,6 +559,18 @@ const struct hda_fixup cs8409_fixups[] = { .chained = true, .chain_id = CS8409_FIXUPS, }, + [CS8409_WARLOCK_MLK] = { + .type = HDA_FIXUP_PINS, + .v.pins = cs8409_cs42l42_pincfgs, + .chained = true, + .chain_id = CS8409_FIXUPS, + }, + [CS8409_WARLOCK_MLK_DUAL_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = cs8409_cs42l42_pincfgs, + .chained = true, + .chain_id = CS8409_FIXUPS, + }, [CS8409_CYBORG] = { .type = HDA_FIXUP_PINS, .v.pins = cs8409_cs42l42_pincfgs, diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 163ff3b3092a..ce5fc03a8065 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -912,9 +912,15 @@ static void cs8409_cs42l42_hw_init(struct hda_codec *codec) cs8409_vendor_coef_set(codec, seq_bullseye->cir, seq_bullseye->coeff); } - /* DMIC1_MO=00b, DMIC1/2_SR=1 */ - if (codec->fixup_id == CS8409_CYBORG) + switch (codec->fixup_id) { + case CS8409_CYBORG: + case CS8409_WARLOCK_MLK_DUAL_MIC: + /* DMIC1_MO=00b, DMIC1/2_SR=1 */ cs8409_vendor_coef_set(codec, CS8409_DMIC_CFG, 0x0003); + break; + default: + break; + } cs42l42_resume(cs42l42); @@ -1008,6 +1014,11 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, spec->scodecs[CS8409_CODEC0]->full_scale_vol = CS42L42_FULL_SCALE_VOL_MINUS6DB; break; + case CS8409_WARLOCK_MLK: + case CS8409_WARLOCK_MLK_DUAL_MIC: + spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; + spec->scodecs[CS8409_CODEC0]->full_scale_vol = CS42L42_FULL_SCALE_VOL_0DB; + break; default: spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; spec->scodecs[CS8409_CODEC0]->full_scale_vol = diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 8e846f292cd0..7df46bd8d2da 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -267,6 +267,8 @@ enum cs8409_coefficient_index_registers { enum { CS8409_BULLSEYE, CS8409_WARLOCK, + CS8409_WARLOCK_MLK, + CS8409_WARLOCK_MLK_DUAL_MIC, CS8409_CYBORG, CS8409_FIXUPS, CS8409_DOLPHIN, -- cgit v1.2.3-58-ga151 From 5e74a144837997e6efc52c56d6686fb6c11c627e Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 28 Mar 2022 12:56:13 +0100 Subject: ALSA: hda/cs8409: Disable HSBIAS_SENSE_EN for Cyborg For ESD reasons, all variants should now set HSBIAS_SENSE_EN. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-6-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index ce5fc03a8065..343fabc4387d 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1005,15 +1005,8 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, cs8409_fix_caps(codec, CS8409_CS42L42_HP_PIN_NID); cs8409_fix_caps(codec, CS8409_CS42L42_AMIC_PIN_NID); - /* Set TIP_SENSE_EN for analog front-end of tip sense. - * Additionally set HSBIAS_SENSE_EN and Full Scale volume for some variants. - */ + /* Set HSBIAS_SENSE_EN and Full Scale volume for some variants. */ switch (codec->fixup_id) { - case CS8409_CYBORG: - spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x00a0; - spec->scodecs[CS8409_CODEC0]->full_scale_vol = - CS42L42_FULL_SCALE_VOL_MINUS6DB; - break; case CS8409_WARLOCK_MLK: case CS8409_WARLOCK_MLK_DUAL_MIC: spec->scodecs[CS8409_CODEC0]->hsbias_hiz = 0x0020; -- cgit v1.2.3-58-ga151 From 5e2baa04e4cd94c2465b248b7d861fcff8c22fae Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Mon, 28 Mar 2022 12:56:14 +0100 Subject: ALSA: hda/cs8409: Add new Dolphin HW variants Add 5 new Dolphin Systems, same configuration as older systems. Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220328115614.15761-7-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 8d20d7fb3d68..74c50ec040d9 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -532,6 +532,11 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BBB, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), SND_PCI_QUIRK(0x1028, 0x0BBC, "Warlock MLK", CS8409_WARLOCK_MLK), SND_PCI_QUIRK(0x1028, 0x0BBD, "Warlock MLK Dual Mic", CS8409_WARLOCK_MLK_DUAL_MIC), + SND_PCI_QUIRK(0x1028, 0x0BD4, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0BD5, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- cgit v1.2.3-58-ga151 From f30741cded62f87bb4b1cc58bc627f076abcaba8 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 30 Mar 2022 14:13:33 +0800 Subject: ALSA: hda/realtek: Fix audio regression on Mi Notebook Pro 2020 Commit 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") is to solve recording issue met on AL236, by matching codec variant ALC269_TYPE_ALC257 and ALC269_TYPE_ALC256. This match can be too broad and Mi Notebook Pro 2020 is broken by the patch. Instead, use codec ID to be narrow down the scope, in order to make ALC256 unaffected. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=215484 Fixes: 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") Reported-by: kernel test robot Reported-by: Dan Carpenter Cc: Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20220330061335.1015533-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c33cb57963d..aace474a899d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3617,8 +3617,8 @@ static void alc256_shutup(struct hda_codec *codec) /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly * when booting with headset plugged. So skip setting it for the codec alc257 */ - if (spec->codec_variant != ALC269_TYPE_ALC257 && - spec->codec_variant != ALC269_TYPE_ALC256) + if (codec->core.vendor_id != 0x10ec0236 && + codec->core.vendor_id != 0x10ec0257) alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) -- cgit v1.2.3-58-ga151 From 6ddc2f749621d5d45ca03edc9f0616bcda136d29 Mon Sep 17 00:00:00 2001 From: Mohan Kumar Date: Tue, 29 Mar 2022 21:29:40 +0530 Subject: ALSA: hda: Avoid unsol event during RPM suspending There is a corner case with unsol event handling during codec runtime suspending state. When the codec runtime suspend call initiated, the codec->in_pm atomic variable would be 0, currently the codec runtime suspend function calls snd_hdac_enter_pm() which will just increments the codec->in_pm atomic variable. Consider unsol event happened just after this step and before snd_hdac_leave_pm() in the codec runtime suspend function. The snd_hdac_power_up_pm() in the unsol event flow in hdmi_present_sense_via_verbs() function would just increment the codec->in_pm atomic variable without calling pm_runtime_get_sync function. As codec runtime suspend flow is already in progress and in parallel unsol event is also accessing the codec verbs, as soon as codec suspend flow completes and clocks are switched off before completing the unsol event handling as both functions doesn't wait for each other. This will result in below errors [ 589.428020] tegra-hda 3510000.hda: azx_get_response timeout, switching to polling mode: last cmd=0x505f2f57 [ 589.428344] tegra-hda 3510000.hda: spurious response 0x80000074:0x5, last cmd=0x505f2f57 [ 589.428547] tegra-hda 3510000.hda: spurious response 0x80000065:0x5, last cmd=0x505f2f57 To avoid this, the unsol event flow should not perform any codec verb related operations during RPM_SUSPENDING state. Signed-off-by: Mohan Kumar Cc: Link: https://lore.kernel.org/r/20220329155940.26331-1-mkumard@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c85ed7bc121e..3e086eebf88d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1625,6 +1625,7 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + struct device *dev = hda_codec_dev(codec); hda_nid_t pin_nid = per_pin->pin_nid; int dev_id = per_pin->dev_id; /* @@ -1638,8 +1639,13 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, int present; int ret; +#ifdef CONFIG_PM + if (dev->power.runtime_status == RPM_SUSPENDING) + return; +#endif + ret = snd_hda_power_up_pm(codec); - if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) + if (ret < 0 && pm_runtime_suspended(dev)) goto out; present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); -- cgit v1.2.3-58-ga151 From bc55cfd5718c7c23e5524582e9fa70b4d10f2433 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Mar 2022 14:09:03 +0200 Subject: ALSA: pcm: Fix potential AB/BA lock with buffer_mutex and mmap_lock syzbot caught a potential deadlock between the PCM runtime->buffer_mutex and the mm->mmap_lock. It was brought by the recent fix to cover the racy read/write and other ioctls, and in that commit, I overlooked a (hopefully only) corner case that may take the revert lock, namely, the OSS mmap. The OSS mmap operation exceptionally allows to re-configure the parameters inside the OSS mmap syscall, where mm->mmap_mutex is already held. Meanwhile, the copy_from/to_user calls at read/write operations also take the mm->mmap_lock internally, hence it may lead to a AB/BA deadlock. A similar problem was already seen in the past and we fixed it with a refcount (in commit b248371628aa). The former fix covered only the call paths with OSS read/write and OSS ioctls, while we need to cover the concurrent access via both ALSA and OSS APIs now. This patch addresses the problem above by replacing the buffer_mutex lock in the read/write operations with a refcount similar as we've used for OSS. The new field, runtime->buffer_accessing, keeps the number of concurrent read/write operations. Unlike the former buffer_mutex protection, this protects only around the copy_from/to_user() calls; the other codes are basically protected by the PCM stream lock. The refcount can be a negative, meaning blocked by the ioctls. If a negative value is seen, the read/write aborts with -EBUSY. In the ioctl side, OTOH, they check this refcount, too, and set to a negative value for blocking unless it's already being accessed. Reported-by: syzbot+6e5c88838328e99c7e1c@syzkaller.appspotmail.com Fixes: dca947d4d26d ("ALSA: pcm: Fix races among concurrent read/write and buffer changes") Cc: Link: https://lore.kernel.org/r/000000000000381a0d05db622a81@google.com Link: https://lore.kernel.org/r/20220330120903.4738-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 9 +++++---- sound/core/pcm_native.c | 39 ++++++++++++++++++++++++++++++++------- 4 files changed, 39 insertions(+), 11 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 314f2779cab5..6b99310b5b88 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,6 +402,7 @@ struct snd_pcm_runtime { struct fasync_struct *fasync; bool stop_operating; /* sync_stop will be called */ struct mutex buffer_mutex; /* protect for buffer changes */ + atomic_t buffer_accessing; /* >0: in r/w operation, <0: blocked */ /* -- private section -- */ void *private_data; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index edd9849210f2..977d54320a5c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -970,6 +970,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, runtime->status->state = SNDRV_PCM_STATE_OPEN; mutex_init(&runtime->buffer_mutex); + atomic_set(&runtime->buffer_accessing, 0); substream->runtime = runtime; substream->private_data = pcm->private_data; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a40a35e51fad..1fc7c50ffa62 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1906,11 +1906,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (avail >= runtime->twake) break; snd_pcm_stream_unlock_irq(substream); - mutex_unlock(&runtime->buffer_mutex); tout = schedule_timeout(wait_time); - mutex_lock(&runtime->buffer_mutex); snd_pcm_stream_lock_irq(substream); set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { @@ -2221,7 +2219,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, nonblock = !!(substream->f_flags & O_NONBLOCK); - mutex_lock(&runtime->buffer_mutex); snd_pcm_stream_lock_irq(substream); err = pcm_accessible_state(runtime); if (err < 0) @@ -2276,6 +2273,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, err = -EINVAL; goto _end_unlock; } + if (!atomic_inc_unless_negative(&runtime->buffer_accessing)) { + err = -EBUSY; + goto _end_unlock; + } snd_pcm_stream_unlock_irq(substream); if (!is_playback) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_CPU); @@ -2284,6 +2285,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (is_playback) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); snd_pcm_stream_lock_irq(substream); + atomic_dec(&runtime->buffer_accessing); if (err < 0) goto _end_unlock; err = pcm_accessible_state(runtime); @@ -2313,7 +2315,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - mutex_unlock(&runtime->buffer_mutex); return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } EXPORT_SYMBOL(__snd_pcm_lib_xfer); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 704fdc9ebf91..4adaee62ef33 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -685,6 +685,24 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, return 0; } +/* acquire buffer_mutex; if it's in r/w operation, return -EBUSY, otherwise + * block the further r/w operations + */ +static int snd_pcm_buffer_access_lock(struct snd_pcm_runtime *runtime) +{ + if (!atomic_dec_unless_positive(&runtime->buffer_accessing)) + return -EBUSY; + mutex_lock(&runtime->buffer_mutex); + return 0; /* keep buffer_mutex, unlocked by below */ +} + +/* release buffer_mutex and clear r/w access flag */ +static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) +{ + mutex_unlock(&runtime->buffer_mutex); + atomic_inc(&runtime->buffer_accessing); +} + #if IS_ENABLED(CONFIG_SND_PCM_OSS) #define is_oss_stream(substream) ((substream)->oss.oss) #else @@ -695,14 +713,16 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime; - int err = 0, usecs; + int err, usecs; unsigned int bits; snd_pcm_uframes_t frames; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - mutex_lock(&runtime->buffer_mutex); + err = snd_pcm_buffer_access_lock(runtime); + if (err < 0) + return err; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: @@ -820,7 +840,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_lib_free_pages(substream); } unlock: - mutex_unlock(&runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(runtime); return err; } @@ -865,7 +885,9 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - mutex_lock(&runtime->buffer_mutex); + result = snd_pcm_buffer_access_lock(runtime); + if (result < 0) + return result; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_SETUP: @@ -884,7 +906,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); unlock: - mutex_unlock(&runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(runtime); return result; } @@ -1369,12 +1391,15 @@ static int snd_pcm_action_nonatomic(const struct action_ops *ops, /* Guarantee the group members won't change during non-atomic action */ down_read(&snd_pcm_link_rwsem); - mutex_lock(&substream->runtime->buffer_mutex); + res = snd_pcm_buffer_access_lock(substream->runtime); + if (res < 0) + goto unlock; if (snd_pcm_stream_linked(substream)) res = snd_pcm_action_group(ops, substream, state, false); else res = snd_pcm_action_single(ops, substream, state); - mutex_unlock(&substream->runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(substream->runtime); + unlock: up_read(&snd_pcm_link_rwsem); return res; } -- cgit v1.2.3-58-ga151