From 41daf6ba594d55f201c50280ebcd430590441da1 Mon Sep 17 00:00:00 2001 From: Kefeng Wang Date: Mon, 24 May 2021 10:49:41 +0800 Subject: ASoC: core: Fix Null-point-dereference in fmt_single_name() Check the return value of devm_kstrdup() in case of Null-point-dereference. Fixes: 45dd9943fce0 ("ASoC: core: remove artificial component and DAI name constraint") Cc: Dmitry Baryshkov Reported-by: Hulk Robot Signed-off-by: Kefeng Wang Link: https://lore.kernel.org/r/20210524024941.159952-1-wangkefeng.wang@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c0904acb935..a76974ccfce1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2225,6 +2225,8 @@ static char *fmt_single_name(struct device *dev, int *id) return NULL; name = devm_kstrdup(dev, devname, GFP_KERNEL); + if (!name) + return NULL; /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); -- cgit v1.2.3 From 6308c44ed6eeadf65c0a7ba68d609773ed860fbb Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Thu, 27 May 2021 01:06:51 +0000 Subject: ASoC: rt5659: Fix the lost powers for the HDA header The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou Signed-off-by: Jack Yu Message-Id: <15fced51977b458798ca4eebf03dafb9@realtek.com> Signed-off-by: Mark Brown --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 87f5709fe2cc..4a50b169fe03 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- cgit v1.2.3 From 4ad7935df6a566225c3d51900bde8f2f0f8b6de3 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 28 May 2021 21:51:23 +0300 Subject: ALSA: hda: Add AlderLake-M PCI ID MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add HD Audio PCI ID for Intel AlderLake-M. Add rules to snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with PCH-DMIC or Soundwire codecs, and legacy driver for the rest. Signed-off-by: Kai Vehmanen Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ab5ff7867eb9..d8be146793ee 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -331,6 +331,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cc, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..470753b36c8a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-M */ + { PCI_DEVICE(0x8086, 0x51cc), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- cgit v1.2.3 From 08a4b904a2a90246aadd6aa2e4f26abca9037385 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 31 May 2021 20:06:33 +0200 Subject: ALSA: hda: Fix a regression in Capture Switch mixer read The recent commit to drop the HDA-specific mute-LED control, e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger"), caused a regression on the mixer element read for "Capture Switch" when it's built from bind controls. The function create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly via snd_hda_gen_add_kctl() instead of add_control(). Although the commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in add_control() as default, this code path fell out from the radar. As a result, now the driver gives -EPERM error because of the lack of the proper access bit at reading "Capture Switch" element value. Fix the regression by setting the access bit properly. Fixes: e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger") BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634 Link: https://lore.kernel.org/r/20210531180633.27831-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b638fc2ef6f7..1f8018f9ce57 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3520,6 +3520,7 @@ static int cap_sw_put(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new cap_sw_temp = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = cap_sw_info, .get = cap_sw_get, .put = cap_sw_put, -- cgit v1.2.3 From 527ff9550682a3d08066a000435ffd8330bdd729 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 31 May 2021 17:37:54 +0100 Subject: ALSA: hda/cirrus: Set Initial DMIC volume to -26 dB Previously this fix was applied only to Bullseye variant laptops, and should be applied to Cyborg and Warlock variants. Fixes: 45b14fe200ba ("ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye") Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20210531163754.136736-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 726507d0b04c..8629e84fef23 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -2206,10 +2206,9 @@ static void cs8409_cs42l42_fixups(struct hda_codec *codec, break; case HDA_FIXUP_ACT_PROBE: - /* Set initial volume on Bullseye to -26 dB */ - if (codec->fixup_id == CS8409_BULLSEYE) - snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID, - HDA_INPUT, 0, 0xff, 0x19); + /* Set initial DMIC volume to -26 dB */ + snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID, + HDA_INPUT, 0, 0xff, 0x19); snd_hda_gen_add_kctl(&spec->gen, NULL, &cs8409_cs42l42_hp_volume_mixer); snd_hda_gen_add_kctl(&spec->gen, -- cgit v1.2.3 From 901be145a46eb79879367d853194346a549e623d Mon Sep 17 00:00:00 2001 From: Carlos M Date: Mon, 31 May 2021 22:20:26 +0200 Subject: ALSA: hda: Fix for mute key LED for HP Pavilion 15-CK0xx For the HP Pavilion 15-CK0xx, with audio subsystem ID 0x103c:0x841c, adding a line in patch_realtek.c to apply the ALC269_FIXUP_HP_MUTE_LED_MIC3 fix activates the mute key LED. Signed-off-by: Carlos M Cc: Link: https://lore.kernel.org/r/20210531202026.35427-1-carlos.marr.pz@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 61a60c420f6f..43e37145eb5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8303,6 +8303,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- cgit v1.2.3 From ce1f25718b2520d0210c24f1e4145d75c5620c9f Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 1 Jun 2021 11:35:06 +0100 Subject: ASoC: topology: Fix spelling mistake "vesion" -> "version" There are spelling mistakes in comments. Fix them. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20210601103506.9477-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 73076d425efb..4893a56208e0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1901,7 +1901,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, * @src: older version of pcm as a source * @pcm: latest version of pcm created from the source * - * Support from vesion 4. User should free the returned pcm manually. + * Support from version 4. User should free the returned pcm manually. */ static int pcm_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *src, @@ -2089,7 +2089,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, * @src: old version of phyical link config as a source * @link: latest version of physical link config created from the source * - * Support from vesion 4. User need free the returned link config manually. + * Support from version 4. User need free the returned link config manually. */ static int link_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_link_config *src, @@ -2400,7 +2400,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, * @src: old version of manifest as a source * @manifest: latest version of manifest created from the source * - * Support from vesion 4. Users need free the returned manifest manually. + * Support from version 4. Users need free the returned manifest manually. */ static int manifest_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_manifest *src, -- cgit v1.2.3 From a8437f05384cb472518ec21bf4fffbe8f0a47378 Mon Sep 17 00:00:00 2001 From: Nicolas Cavallari Date: Thu, 27 May 2021 18:34:09 +0200 Subject: ASoC: fsl-asoc-card: Set .owner attribute when registering card. Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [] (unwind_backtrace) from [] (show_stack+0x10/0x14) [] (show_stack) from [] (dump_stack+0xdc/0x104) [] (dump_stack) from [] (__warn+0xd8/0x114) [] (__warn) from [] (warn_slowpath_fmt+0x5c/0xc4) [] (warn_slowpath_fmt) from [] (snd_card_new+0x310/0x39c [snd]) [] (snd_card_new [snd]) from [] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [] (snd_soc_bind_card [snd_soc_core]) from [] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [] (devm_snd_soc_register_card [snd_soc_core]) from [] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20210527163409.22049-1-nicolas.cavallari@green-communications.fr Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c62bfd1c3ac7..4f55b316cf0f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- cgit v1.2.3 From b640e8a4bd24e17ce24a064d704aba14831651a8 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 28 May 2021 17:43:30 +0300 Subject: ASoC: SOF: reset enabled_cores state at suspend The recent changes to use common code to power up/down DSP cores also removed the reset of the core state at suspend. It turns out this is still needed. When the firmware state is reset to SOF_FW_BOOT_NOT_STARTED, also enabled_cores should be reset, and existing DSP drivers depend on this. BugLink: https://github.com/thesofproject/linux/issues/2824 Fixes: 42077f08b3 ("ASoC: SOF: update dsp core power status in common APIs") Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20210528144330.2551-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index fd265803f7bc..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -256,6 +256,7 @@ suspend: /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; + sdev->enabled_cores_mask = 0; return ret; } -- cgit v1.2.3 From 3ae72f6ab9c1f688bd578cdc252dabce65fdaf57 Mon Sep 17 00:00:00 2001 From: Dongliang Mu Date: Wed, 2 Jun 2021 11:41:36 +0800 Subject: ALSA: control led: fix memory leak in snd_ctl_led_register The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails to decrease the refcount to zero, which causes device_release never to be invoked. This leads to memory leak to some resources, like struct device_private. In addition, we also free some other similar memory leaks in snd_ctl_led_init/snd_ctl_led_exit. Fix this by replacing device_del to device_unregister in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit. Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will call kobject_release and delay the release of kobject, which will cause use-after-free when the memory backing the kobject is freed at once. Reported-by: syzbot+08a7d8b51ea048a74ffb@syzkaller.appspotmail.com Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Signed-off-by: Dongliang Mu Reviewed-by: Dan Carpenter Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20210602034136.2762497-1-mudongliangabcd@gmail.com Signed-off-by: Takashi Iwai --- sound/core/control_led.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a90e31dbde61 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -17,6 +17,9 @@ MODULE_LICENSE("GPL"); #define MAX_LED (((SNDRV_CTL_ELEM_ACCESS_MIC_LED - SNDRV_CTL_ELEM_ACCESS_SPK_LED) \ >> SNDRV_CTL_ELEM_ACCESS_LED_SHIFT) + 1) +#define to_led_card_dev(_dev) \ + container_of(_dev, struct snd_ctl_led_card, dev) + enum snd_ctl_led_mode { MODE_FOLLOW_MUTE = 0, MODE_FOLLOW_ROUTE, @@ -371,6 +374,21 @@ static void snd_ctl_led_disconnect(struct snd_card *card) snd_ctl_led_refresh(); } +static void snd_ctl_led_card_release(struct device *dev) +{ + struct snd_ctl_led_card *led_card = to_led_card_dev(dev); + + kfree(led_card); +} + +static void snd_ctl_led_release(struct device *dev) +{ +} + +static void snd_ctl_led_dev_release(struct device *dev) +{ +} + /* * sysfs */ @@ -663,6 +681,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) led_card->number = card->number; led_card->led = led; device_initialize(&led_card->dev); + led_card->dev.release = snd_ctl_led_card_release; if (dev_set_name(&led_card->dev, "card%d", card->number) < 0) goto cerr; led_card->dev.parent = &led->dev; @@ -681,7 +700,6 @@ cerr: put_device(&led_card->dev); cerr2: printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number); - kfree(led_card); } } @@ -700,8 +718,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) snprintf(link_name, sizeof(link_name), "led-%s", led->name); sysfs_remove_link(&card->ctl_dev.kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); - device_del(&led_card->dev); - kfree(led_card); + device_unregister(&led_card->dev); led->cards[card->number] = NULL; } } @@ -723,6 +740,7 @@ static int __init snd_ctl_led_init(void) device_initialize(&snd_ctl_led_dev); snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { put_device(&snd_ctl_led_dev); @@ -733,15 +751,16 @@ static int __init snd_ctl_led_init(void) INIT_LIST_HEAD(&led->controls); device_initialize(&led->dev); led->dev.parent = &snd_ctl_led_dev; + led->dev.release = snd_ctl_led_release; led->dev.groups = snd_ctl_led_dev_attr_groups; dev_set_name(&led->dev, led->name); if (device_add(&led->dev)) { put_device(&led->dev); for (; group > 0; group--) { led = &snd_ctl_leds[group - 1]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); return -ENOMEM; } } @@ -767,9 +786,9 @@ static void __exit snd_ctl_led_exit(void) } for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); snd_ctl_led_clean(NULL); } -- cgit v1.2.3 From 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d Mon Sep 17 00:00:00 2001 From: Mark Pearson Date: Mon, 31 May 2021 10:55:02 -0400 Subject: ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu Signed-off-by: Mark Pearson Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..c9fb1c8fbf8c 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- cgit v1.2.3 From 9c1fe96bded935369f8340c2ac2e9e189f697d5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Jun 2021 13:38:23 +0200 Subject: ALSA: timer: Fix master timer notification snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e@syzkaller.appspotmail.com Reviewed-by: Jaroslav Kysela Cc: Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Link: https://lore.kernel.org/r/20210602113823.23777-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- cgit v1.2.3 From b8b90c17602689eeaa5b219d104bbc215d1225cc Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 2 Jun 2021 22:54:24 +0800 Subject: ALSA: hda: update the power_state during the direct-complete The patch_realtek.c needs to check if the power_state.event equals PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and resume() will be skipped if the codec is already rt_suspended, in this case, the patch_realtek.c will always get PM_EVENT_ON even the system is really resumed from S3. We could set power_state to PMSG_SUSPEND in the prepare(), if other PM functions are called before complete(), those functions will override power_state; if no other PM functions are called before complete(), we could know the suspend() and resume() are skipped since only S3 pm functions could be skipped by direct-complete, in this case set power_state to PMSG_RESUME in the complete(). This could guarantee the first time of calling hda_codec_runtime_resume() after complete() has the correct power_state. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20210602145424.3132-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a31009afc025..5462f771c2f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2917,6 +2917,7 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2924,6 +2925,10 @@ static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* If no other pm-functions are called between prepare() and complete() */ + if (dev->power.power_state.event == PM_EVENT_SUSPEND) + dev->power.power_state = PMSG_RESUME; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || hda_codec_need_resume(codec) || codec->forced_resume)) pm_request_resume(dev); -- cgit v1.2.3 From 320232caf1d8febea17312dab4b2dfe02e033520 Mon Sep 17 00:00:00 2001 From: Mark Pearson Date: Wed, 2 Jun 2021 13:12:51 -0400 Subject: ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu Signed-off-by: Mark Pearson Link: https://lore.kernel.org/r/20210602171251.3243-1-markpearson@lenovo.com Signed-off-by: Mark Brown --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------------- 1 file changed, 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index c9fb1c8fbf8c..19438da5dfa5 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- cgit v1.2.3 From 8bef925e37bdc9b6554b85eda16ced9a8e3c135f Mon Sep 17 00:00:00 2001 From: Richard Weinberger Date: Sun, 30 May 2021 22:34:46 +0200 Subject: ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger Link: https://lore.kernel.org/r/20210530203446.19022-1-richard@nod.at Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- cgit v1.2.3 From 49783c6f4a4f49836b5a109ae0daf2f90b0d7713 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Fri, 4 Jun 2021 14:31:50 +0800 Subject: ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20210604063150.29925-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index fed80c8f994f..e78ba3b064c4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- cgit v1.2.3 From 15d295b560e6dd45f839a53ae69e4f63b54eb32f Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Sat, 5 Jun 2021 16:25:36 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2 The HP Elite Dragonfly G2 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. In the other hand, there is no output from right channel of speaker. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20210605082539.41797-1-jeremy.szu@canonical.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 43e37145eb5d..9f65171a902d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8310,6 +8310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), + SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- cgit v1.2.3 From 61d3e87468fad82dc8e8cb6de7db563ada64b532 Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Sat, 5 Jun 2021 16:25:37 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP EliteBook x360 1040 G8 The HP EliteBook x360 1040 G8 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. In the other hand, there is no output from right channel of speaker. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20210605082539.41797-2-jeremy.szu@canonical.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 9f65171a902d..11324163ebe1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8311,6 +8311,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- cgit v1.2.3 From dfb06401b4cdfc71e2fc3e19b877ab845cc9f7f7 Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Sat, 5 Jun 2021 16:25:38 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 840 Aero G8 The HP EliteBook 840 Aero G8 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. In the other hand, there is no output from right channel of speaker. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20210605082539.41797-3-jeremy.szu@canonical.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 11324163ebe1..215beb3ac678 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8330,6 +8330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- cgit v1.2.3 From 9981b20a5e3694f4625ab5a1ddc98ce7503f6d12 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 5 Jun 2021 18:10:54 +0900 Subject: ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun() In the workqueue to queue wake-up event, isochronous context is not processed, thus it's useless to check context for the workqueue to switch status of runtime for PCM substream to XRUN. On the other hand, in software IRQ context of 1394 OHCI, it's needed. This commit fixes the bug introduced when tasklet was replaced with workqueue. Cc: Fixes: 2b3d2987d800 ("ALSA: firewire: Replace tasklet with work") Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20210605091054.68866-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e0faa6601966..5805c5de39fb 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -804,7 +804,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (current_work() == &s->period_work) + if (in_interrupt()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } -- cgit v1.2.3 From c8a4556d98510ca05bad8d02265a4918b03a8c0b Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Fri, 4 Jun 2021 23:45:45 +0800 Subject: ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) Co-developed-by: Judy Hsiao Signed-off-by: Judy Hsiao Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210604154545.1198337-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 +++ 2 files changed, 83 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 28c7497344e3..a6e95db6b3fb 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- cgit v1.2.3 From 57c9e21a49b1c196cda28f54de9a5d556ac93f20 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 8 Jun 2021 10:46:00 +0800 Subject: ALSA: hda/realtek: headphone and mic don't work on an Acer laptop There are 2 issues on this machine, the 1st one is mic's plug/unplug can't be detected, that is because the mic is set to manual detecting mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto detecting mode. The other one is headphone's plug/unplug can't be detected by pulseaudio, that is because the pulseaudio will use ucm2/sof-hda-dsp on this machine, and the ucm2 only handle 'Headphone Jack', but on this machine the headphone's pincfg sets the location to Front, then the alsa mixer name is "Front Headphone Jack" instead of "Headphone Jack", so override the pincfg to change location to Left. BugLink: http://bugs.launchpad.net/bugs/1930188 Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20210608024600.6198-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 215beb3ac678..11ba8e351ad4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,7 @@ enum { ALC285_FIXUP_HP_SPECTRE_X360, ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, + ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC283_FIXUP_HEADSET_MIC, }, + [ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x21, 0x03211030 }, /* Change the Headphone location to Left */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -8740,6 +8751,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, + {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {} }; #define ALC225_STANDARD_PINS \ -- cgit v1.2.3 From 600dd2a7e8b62170d177381cc1303861f48f9780 Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Tue, 8 Jun 2021 19:47:48 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8 The HP ZBook Power G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20210608114750.32009-1-jeremy.szu@canonical.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 11ba8e351ad4..ab5113cccffa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8346,6 +8346,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- cgit v1.2.3 From 83e197a8414c0ba545e7e3916ce05f836f349273 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Jun 2021 17:20:59 +0200 Subject: ALSA: seq: Fix race of snd_seq_timer_open() The timer instance per queue is exclusive, and snd_seq_timer_open() should have managed the concurrent accesses. It looks as if it's checking the already existing timer instance at the beginning, but it's not right, because there is no protection, hence any later concurrent call of snd_seq_timer_open() may override the timer instance easily. This may result in UAF, as the leftover timer instance can keep running while the queue itself gets closed, as spotted by syzkaller recently. For avoiding the race, add a proper check at the assignment of tmr->timeri again, and return -EBUSY if it's been already registered. Reported-by: syzbot+ddc1260a83ed1cbf6fb5@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com Link: https://lore.kernel.org/r/20210610152059.24633-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_timer.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 1645e4142e30..9863be6fd43e 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q) return err; } spin_lock_irq(&tmr->lock); - tmr->timeri = t; + if (tmr->timeri) + err = -EBUSY; + else + tmr->timeri = t; spin_unlock_irq(&tmr->lock); + if (err < 0) { + snd_timer_close(t); + snd_timer_instance_free(t); + return err; + } return 0; } -- cgit v1.2.3