From cdf72837cda89b2d38bd18fbe6cc591c1d5f2416 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Fri, 23 Jul 2021 05:41:53 +0930 Subject: ALSA: scarlett2: Fix Mute/Dim/MSD Mode control names Append "Playback Switch" to the names of "Mute" and "Dim" controls, and append "Switch" to the "MSD Mode" control as per Documentation/sound/designs/control-names.rst. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/77f1000652c37e3217fb8dad8e156bc6392abc0b.1626959758.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index f9d698a37153..347995ea39e4 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -228,7 +228,7 @@ enum { }; static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { - "Mute", "Dim" + "Mute Playback Switch", "Dim Playback Switch" }; /* Description of each hardware port type: @@ -3455,7 +3455,7 @@ static int scarlett2_add_msd_ctl(struct usb_mixer_interface *mixer) /* Add MSD control */ return scarlett2_add_new_ctl(mixer, &scarlett2_msd_ctl, - 0, 1, "MSD Mode", NULL); + 0, 1, "MSD Mode Switch", NULL); } /*** Cleanup/Suspend Callbacks ***/ -- cgit v1.2.3-59-g8ed1b From d3a4f784d20c696b134b916f57956f12a37ecd47 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Fri, 23 Jul 2021 05:42:08 +0930 Subject: ALSA: scarlett2: Fix Direct Monitor control name for 2i2 The Direct Monitor control for the 2i2 is an enumerated value, not a boolean. Fix the control name to say "Playback Enum" instead of "Playback Switch" in this case. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/faf5de1d2100038e7d07520d770fda4a1adc276a.1626959758.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 347995ea39e4..fa604b61066f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2530,14 +2530,18 @@ static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; + const char *s; if (!info->direct_monitor) return 0; + s = info->direct_monitor == 1 + ? "Direct Monitor Playback Switch" + : "Direct Monitor Playback Enum"; + return scarlett2_add_new_ctl( mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1], - 0, 1, "Direct Monitor Playback Switch", - &private->direct_monitor_ctl); + 0, 1, s, &private->direct_monitor_ctl); } /*** Speaker Switching Control ***/ -- cgit v1.2.3-59-g8ed1b From 9ee0fc8366ddce380547878640708f1bd7dd2ead Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Fri, 23 Jul 2021 05:42:48 +0930 Subject: ALSA: scarlett2: Correct channel mute status after mute button pressed After the hardware mute button is pressed, private->vol_updated is set so that the mute status is invalidated. As the channel mute values may be affected by the global mute value, update scarlett2_mute_ctl_get() to call scarlett2_update_volumes() if private->vol_updated is set. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/aa18ddbf8d8bd7f31832ab1b6b6057c00b931202.1626959758.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index fa604b61066f..3457fbc8108f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1856,9 +1856,15 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_data *private = elem->head.mixer->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; int index = line_out_remap(private, elem->control); + mutex_lock(&private->data_mutex); + if (private->vol_updated) + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + ucontrol->value.integer.value[0] = private->mute_switch[index]; return 0; } -- cgit v1.2.3-59-g8ed1b From 2b8b12be9b9752c36efda38b7dd5d83d790d01d8 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Fri, 23 Jul 2021 05:43:26 +0930 Subject: ALSA: scarlett2: Fix line out/speaker switching notifications The values of the line output controls can change when the SW/HW switches are set to HW, and also when speaker switching is enabled. These notifications were sent with a mask of only SNDRV_CTL_EVENT_MASK_INFO. Change the notifications to set the SNDRV_CTL_EVENT_MASK_VALUE mask bit as well. When the mute control is updated, the notification was sent with a mask of SNDRV_CTL_EVENT_MASK_INFO. Change the mask to the correct value of SNDRV_CTL_EVENT_MASK_VALUE. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/8192e15ba62fa4bc90425c005f265c0de530be20.1626959758.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 3457fbc8108f..3d5848d5481b 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1961,10 +1961,12 @@ static void scarlett2_vol_ctl_set_writable(struct usb_mixer_interface *mixer, ~SNDRV_CTL_ELEM_ACCESS_WRITE; } - /* Notify of write bit change */ - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + /* Notify of write bit and possible value change */ + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &private->vol_ctls[index]->id); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &private->mute_ctls[index]->id); } @@ -2599,7 +2601,9 @@ static int scarlett2_speaker_switch_enable(struct usb_mixer_interface *mixer) /* disable the line out SW/HW switch */ scarlett2_sw_hw_ctl_ro(private, i); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &private->sw_hw_ctls[i]->id); } @@ -2923,7 +2927,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, if (private->vol_sw_hw_switch[line_index]) { private->mute_switch[line_index] = val; snd_ctl_notify(mixer->chip->card, - SNDRV_CTL_EVENT_MASK_INFO, + SNDRV_CTL_EVENT_MASK_VALUE, &private->mute_ctls[i]->id); } } -- cgit v1.2.3-59-g8ed1b From 4511781f95da0a3b2bad34f3f5e3967e80cd2d18 Mon Sep 17 00:00:00 2001 From: "chihhao.chen" Date: Sat, 24 Jul 2021 12:23:41 +0800 Subject: ALSA: usb-audio: fix incorrect clock source setting The following scenario describes an echo test for Samsung USBC Headset (AKG) with VID/PID (0x04e8/0xa051). We first start a capture stream(USB IN transfer) in 96Khz/24bit/1ch mode. In clock find source function, we get value 0x2 for clock selector and 0x1 for clock source. Kernel-4.14 behavior Since clock source is valid so clock selector was not set again. We pass through this function and start a playback stream(USB OUT transfer) in 48Khz/32bit/2ch mode. This time we get value 0x1 for clock selector and 0x1 for clock source. Finally clock id with this setting is 0x9. Kernel-5.10 behavior Clock selector was always set one more time even it is valid. When we start a playback stream, we will get 0x2 for clock selector and 0x1 for clock source. In this case clock id becomes 0xA. This is an incorrect clock source setting and results in severe noises. We see wrong data rate in USB IN transfer. (From 288 bytes/ms becomes 144 bytes/ms) It should keep in 288 bytes/ms. This earphone works fine on older kernel version load because this is a newly-added behavior. Fixes: d2e8f641257d ("ALSA: usb-audio: Explicitly set up the clock selector") Signed-off-by: chihhao.chen Link: https://lore.kernel.org/r/1627100621-19225-1-git-send-email-chihhao.chen@mediatek.com Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 52de52288e10..14456f61539e 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -324,6 +324,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, sources[ret - 1], visited, validate); if (ret > 0) { + /* + * For Samsung USBC Headset (AKG), setting clock selector again + * will result in incorrect default clock setting problems + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051)) + return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) return err; -- cgit v1.2.3-59-g8ed1b From 66291b6adb66dd3bc96b0f594d88c2ff1300d95f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Jul 2021 08:26:56 +0200 Subject: ALSA: usb-audio: Fix superfluous autosuspend recovery The change to restore the autosuspend from the disabled state uses a wrong check: namely, it should have been the exact comparison of the quirk_type instead of the bitwise and (&). Otherwise it matches wrongly with the other quirk types. Although re-enabling the autosuspend for the already enabled device shouldn't matter much, it's better to fix the unbalanced call. Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()") Cc: Link: https://lore.kernel.org/r/s5hr1flh9ov.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..a1f8c3a026f5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) + if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND) usb_enable_autosuspend(interface_to_usbdev(intf)); chip->num_interfaces--; -- cgit v1.2.3-59-g8ed1b From 35171fbfc0d94aa31b009bb475d156ad1941ab50 Mon Sep 17 00:00:00 2001 From: Nikos Liolios Date: Tue, 27 Jul 2021 06:05:10 +0300 Subject: ALSA: hda/realtek: Fix headset mic for Acer SWIFT SF314-56 (ALC256) The issue on Acer SWIFT SF314-56 is that headset microphone doesn't work. The following quirk fixed headset microphone issue. The fixup was found by trial and error. Note that the fixup of SF314-54/55 (ALC256_FIXUP_ACER_HEADSET_MIC) was not successful on my SF314-56. Signed-off-by: Nikos Liolios Cc: Link: https://lore.kernel.org/r/20210727030510.36292-1-liolios.nk@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 caaf0e8aac11..14e1ab7c7954 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8274,6 +8274,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 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), -- cgit v1.2.3-59-g8ed1b From 4b0556b96e1fe7723629bd40e3813a30cd632faf Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Tue, 27 Jul 2021 12:33:26 +0300 Subject: ALSA: usb-audio: Add registration quirk for JBL Quantum 600 Apparently JBL Quantum 600 has multiple hardware revisions. Apply registration quirk to another device id as well. Signed-off-by: Alexander Tsoy Cc: Link: https://lore.kernel.org/r/20210727093326.1153366-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e7accd87e063..326d1b0ea5e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1899,6 +1899,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; -- cgit v1.2.3-59-g8ed1b From 0d4867a185460397af56b9afe3e2243d3e610e37 Mon Sep 17 00:00:00 2001 From: Alexander Monakov Date: Wed, 21 Jul 2021 20:01:41 +0300 Subject: ALSA: hda/realtek: add mic quirk for Acer SF314-42 The Acer Swift SF314-42 laptop is using Realtek ALC255 codec. Add a quirk so microphone in a headset connected via the right-hand side jack is usable. Signed-off-by: Alexander Monakov Cc: Link: https://lore.kernel.org/r/20210721170141.24807-1-amonakov@ispras.ru 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 14e1ab7c7954..21c521596c9d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8278,6 +8278,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 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, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 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), -- cgit v1.2.3-59-g8ed1b From 8dde723fcde4479f256441da03793e37181d9f21 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 Jul 2021 20:51:26 +0200 Subject: ALSA: usb-audio: Avoid unnecessary or invalid connector selection at resume The recent fix for the resume on Lenovo machines seems causing a regression on others. It's because the change always triggers the connector selection no matter which widget node type is. This patch addresses the regression by setting the resume callback selectively only for the connector widget. Fixes: 44609fc01f28 ("ALSA: usb-audio: Check connector value on resume") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897 Link: https://lore.kernel.org/r/20210729185126.24432-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..9b713b4a5ec4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, strlcat(name, " - Output Jack", name_size); } +/* get connector value to "wake up" the USB audio */ +static int connector_mixer_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); + + get_connector_value(cval, NULL, NULL); + return 0; +} + /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, @@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); + + /* set up a specific resume callback */ + cval->head.resume = connector_mixer_resume; + /* * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the * number of channels connected. @@ -3642,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_resume(struct usb_mixer_elem_list *list) -{ - struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - - /* get connector value to "wake up" the USB audio */ - if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1) - get_connector_value(cval, NULL, NULL); - - return 0; -} - static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) { - int err = default_mixer_resume(list); + int err; - if (err < 0) - return err; + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } return restore_mixer_value(list); } @@ -3697,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = default_mixer_resume; + list->resume = NULL; list->reset_resume = default_mixer_reset_resume; #endif } -- cgit v1.2.3-59-g8ed1b From 852a8a97776a153be2e6c803218eced45f37a19c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 30 Jul 2021 11:02:54 +0200 Subject: ALSA: pcm - fix mmap capability check for the snd-dummy driver The snd-dummy driver (fake_buffer configuration) uses the ops->page callback for the mmap operations. Allow mmap for this case, too. Cc: Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check") Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20210730090254.612478-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6a2971a7e6a1..09c0e2a6489c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,7 +246,7 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap) + if (substream->ops->mmap || substream->ops->page) return true; switch (substream->dma_buffer.dev.type) { -- cgit v1.2.3-59-g8ed1b From eda80d7c9c4db0f55f130e38c682e19b58d5add7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 1 Aug 2021 13:38:01 +0200 Subject: ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS The recent code refactoring made the mmap of continuous pages to be done via the own helper snd_dma_continuous_mmap() with remap_pfn_range(). There I overlooked that dmab->addr isn't set for the allocation with SNDRV_DMA_TYPE_CONTINUOUS. This resulted always in an error at mmap with this buffer type on the system such as Intel SST Baytrail driver. This patch fixes the regression by passing the correct address. Fixes: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") Reported-by: Hans de Goede Link: https://lore.kernel.org/r/8d6674da-7d7b-803e-acc9-7de6cb1223fa@redhat.com Link: https://lore.kernel.org/r/20210801113801.31290-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 83b79edfa52d..439a358ecfe9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -215,7 +215,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - dmab->addr >> PAGE_SHIFT, + page_to_pfn(virt_to_page(dmab->area)), area->vm_end - area->vm_start, area->vm_page_prot); } -- cgit v1.2.3-59-g8ed1b From 97367c97226aab8b298ada954ce12659ee3ad2a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Aug 2021 13:43:12 +0200 Subject: ALSA: seq: Fix racy deletion of subscriber It turned out that the current implementation of the port subscription is racy. The subscription contains two linked lists, and we have to add to or delete from both lists. Since both connection and disconnection procedures perform the same order for those two lists (i.e. src list, then dest list), when a deletion happens during a connection procedure, the src list may be deleted before the dest list addition completes, and this may lead to a use-after-free or an Oops, even though the access to both lists are protected via mutex. The simple workaround for this race is to change the access order for the disconnection, namely, dest list, then src list. This assures that the connection has been established when disconnecting, and also the concurrent deletion can be avoided. Reported-and-tested-by: folkert Cc: Link: https://lore.kernel.org/r/20210801182754.GP890690@belle.intranet.vanheusden.com Link: https://lore.kernel.org/r/20210803114312.2536-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ports.c | 39 +++++++++++++++++++++++++++------------ 1 file changed, 27 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index b9c2ce2b8d5a..84d78630463e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -514,10 +514,11 @@ static int check_and_subscribe_port(struct snd_seq_client *client, return err; } -static void delete_and_unsubscribe_port(struct snd_seq_client *client, - struct snd_seq_client_port *port, - struct snd_seq_subscribers *subs, - bool is_src, bool ack) +/* called with grp->list_mutex held */ +static void __delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) { struct snd_seq_port_subs_info *grp; struct list_head *list; @@ -525,7 +526,6 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp = is_src ? &port->c_src : &port->c_dest; list = is_src ? &subs->src_list : &subs->dest_list; - down_write(&grp->list_mutex); write_lock_irq(&grp->list_lock); empty = list_empty(list); if (!empty) @@ -535,6 +535,18 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); +} + +static void delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) +{ + struct snd_seq_port_subs_info *grp; + + grp = is_src ? &port->c_src : &port->c_dest; + down_write(&grp->list_mutex); + __delete_and_unsubscribe_port(client, port, subs, is_src, ack); up_write(&grp->list_mutex); } @@ -590,27 +602,30 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, struct snd_seq_client_port *dest_port, struct snd_seq_port_subscribe *info) { - struct snd_seq_port_subs_info *src = &src_port->c_src; + struct snd_seq_port_subs_info *dest = &dest_port->c_dest; struct snd_seq_subscribers *subs; int err = -ENOENT; - down_write(&src->list_mutex); + /* always start from deleting the dest port for avoiding concurrent + * deletions + */ + down_write(&dest->list_mutex); /* look for the connection */ - list_for_each_entry(subs, &src->list_head, src_list) { + list_for_each_entry(subs, &dest->list_head, dest_list) { if (match_subs_info(info, &subs->info)) { - atomic_dec(&subs->ref_count); /* mark as not ready */ + __delete_and_unsubscribe_port(dest_client, dest_port, + subs, false, + connector->number != dest_client->number); err = 0; break; } } - up_write(&src->list_mutex); + up_write(&dest->list_mutex); if (err < 0) return err; delete_and_unsubscribe_port(src_client, src_port, subs, true, connector->number != src_client->number); - delete_and_unsubscribe_port(dest_client, dest_port, subs, false, - connector->number != dest_client->number); kfree(subs); return 0; } -- cgit v1.2.3-59-g8ed1b