From 2e2832562c877e6530b8480982d99a4ff90c6777 Mon Sep 17 00:00:00 2001
From: Alan Young <consult.awy@gmail.com>
Date: Fri, 9 Jul 2021 09:48:54 +0100
Subject: ALSA: pcm: Call substream ack() method upon compat mmap commit
If a 32-bit application is being used with a 64-bit kernel and is using
the mmap mechanism to write data, then the SNDRV_PCM_IOCTL_SYNC_PTR
ioctl results in calling snd_pcm_ioctl_sync_ptr_compat(). Make this use
pcm_lib_apply_appl_ptr() so that the substream's ack() method, if
defined, is called.
The snd_pcm_sync_ptr() function, used in the 64-bit ioctl case, already
uses snd_pcm_ioctl_sync_ptr_compat().
Fixes: 9027c4639ef1 ("ALSA: pcm: Call ack() whenever appl_ptr is updated")
Signed-off-by: Alan Young <consult.awy@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/c441f18c-eb2a-3bdd-299a-696ccca2de9c@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 11 ++++++++---
1 file changed, 8 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 14e32825c339..c88c4316c417 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3063,9 +3063,14 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream,
boundary = 0x7fffffff;
snd_pcm_stream_lock_irq(substream);
/* FIXME: we should consider the boundary for the sync from app */
- if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL))
- control->appl_ptr = scontrol.appl_ptr;
- else
+ if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) {
+ err = pcm_lib_apply_appl_ptr(substream,
+ scontrol.appl_ptr);
+ if (err < 0) {
+ snd_pcm_stream_unlock_irq(substream);
+ return err;
+ }
+ } else
scontrol.appl_ptr = control->appl_ptr % boundary;
if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN))
control->avail_min = scontrol.avail_min;
--
cgit
From 4796372e7c117b84bfd929526f48e23c79030dcd Mon Sep 17 00:00:00 2001
From: "Gustavo A. R. Silva" <gustavoars@kernel.org>
Date: Tue, 13 Jul 2021 14:58:18 -0500
Subject: ASoC: Mediatek: MT8183: Fix fall-through warning for Clang
Fix the following fallthrough warning:
sound/soc/mediatek/mt8183/mt8183-dai-adda.c:342:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough]
Reported-by: Nathan Chancellor <nathan@kernel.org>
Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org>
---
sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
index 2b758a18c2ea..5b8a274419ed 100644
--- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
+++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c
@@ -341,6 +341,7 @@ static int set_mtkaif_rx(struct mtk_base_afe *afe)
case MT8183_MTKAIF_PROTOCOL_1:
regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31);
regmap_write(afe->regmap, AFE_ADDA_MTKAIF_CFG0, 0x0);
+ break;
default:
break;
}
--
cgit
From 64752a95b702817602d72f109ceaf5ec0780e283 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Jul 2021 10:48:36 +0200
Subject: ALSA: usb-audio: Add missing proc text entry for BESPOKEN type
Recently we've added a new usb_mixer element type, USB_MIXER_BESPOKEN,
but it wasn't added in the table in snd_usb_mixer_dump_cval(). This
is no big problem since each bespoken type should have its own dump
method, but it still isn't disallowed to use the standard one, so we
should cover it as well. Along with it, define the table with the
explicit array initializer for avoiding other pitfalls.
Fixes: 785b6f29a795 ("ALSA: usb-audio: scarlett2: Fix wrong resume call")
Reported-by: Pavel Machek <pavel@denx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210714084836.1977-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/mixer.c | 10 +++++++++-
1 file changed, 9 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 30b3e128e28d..f4cdaf1ba44a 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -3295,7 +3295,15 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
{
struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
static const char * const val_types[] = {
- "BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32",
+ [USB_MIXER_BOOLEAN] = "BOOLEAN",
+ [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN",
+ [USB_MIXER_S8] = "S8",
+ [USB_MIXER_U8] = "U8",
+ [USB_MIXER_S16] = "S16",
+ [USB_MIXER_U16] = "U16",
+ [USB_MIXER_S32] = "S32",
+ [USB_MIXER_U32] = "U32",
+ [USB_MIXER_BESPOKEN] = "BESPOKEN",
};
snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, "
"channels=%i, type=\"%s\"\n", cval->head.id,
--
cgit
From 1c2b9519159b470ef24b2638f4794e86e2952ab7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 16 Jul 2021 15:27:23 +0200
Subject: ALSA: sb: Fix potential ABBA deadlock in CSP driver
SB16 CSP driver may hit potentially a typical ABBA deadlock in two
code paths:
In snd_sb_csp_stop():
spin_lock_irqsave(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
In snd_sb_csp_load():
spin_lock_irqsave(&p->chip->reg_lock, flags);
spin_lock(&p->chip->mixer_lock);
Also the similar pattern is seen in snd_sb_csp_start().
Although the practical impact is very small (those states aren't
triggered in the same running state and this happens only on a real
hardware, decades old ISA sound boards -- which must be very difficult
to find nowadays), it's a real scenario and has to be fixed.
This patch addresses those deadlocks by splitting the locks in
snd_sb_csp_start() and snd_sb_csp_stop() for avoiding the nested
locks.
Reported-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/7b0fcdaf-cd4f-4728-2eae-48c151a92e10@gmail.com
Link: https://lore.kernel.org/r/20210716132723.13216-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/isa/sb/sb16_csp.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 5bbe6695689d..7ad8c5f7b664 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -816,6 +816,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
set_mode_register(p->chip, 0xc0); /* c0 = STOP */
@@ -855,6 +856,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@@ -880,6 +882,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
@@ -894,6 +897,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p)
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
--
cgit
From 33f735f137c6539e3ceceb515cd1e2a644005b49 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 16 Jul 2021 15:56:00 +0200
Subject: ALSA: hdmi: Expose all pins on MSI MS-7C94 board
The BIOS on MSI Mortar B550m WiFi (MS-7C94) board with AMDGPU seems
disabling the other pins than HDMI although it has more outputs
including DP.
This patch adds the board to the allow list for enabling all pins.
Reported-by: Damjan Georgievski <gdamjan@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAEk1YH4Jd0a8vfZxORVu7qg+Zsc-K+pR187ezNq8QhJBPW4gpw@mail.gmail.com
Link: https://lore.kernel.org/r/20210716135600.24176-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_hdmi.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 4b2cc8cb55c4..84c088912b3c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1940,6 +1940,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
static const struct snd_pci_quirk force_connect_list[] = {
SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+ SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
{}
};
--
cgit
From e4efa82660e6d80338c554e45e903714e1b2c27b Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Mon, 19 Jul 2021 11:02:31 +0800
Subject: ALSA: hda/realtek: Fix pop noise and 2 Front Mic issues on a machine
This is a Lenovo ThinkStation machine which uses the codec alc623.
There are 2 issues on this machine, the 1st one is the pop noise in
the lineout, the 2nd one is there are 2 Front Mics and pulseaudio
can't handle them, After applying the fixup of
ALC623_FIXUP_LENOVO_THINKSTATION_P340 to this machine, the 2 issues
are fixed.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20210719030231.6870-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 1389cfd5e0db..caaf0e8aac11 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8626,6 +8626,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340),
SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
--
cgit
From 114613f62f42e7cbc1242c4e82076a0153043761 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Mon, 19 Jul 2021 18:17:46 -0500
Subject: ALSA: hda: intel-dsp-cfg: add missing ElkhartLake PCI ID
We missed the fact that ElkhartLake platforms have two different PCI
IDs. We only added one so the SOF driver is never selected by the
autodetection logic for the missing configuration.
BugLink: https://github.com/thesofproject/linux/issues/2990
Fixes: cc8f81c7e625 ('ALSA: hda: fix intel DSP config')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210719231746.557325-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/hda/intel-dsp-config.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index d8be146793ee..c9d0ba353463 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -319,6 +319,10 @@ static const struct config_entry config_table[] = {
.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
.device = 0x4b55,
},
+ {
+ .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+ .device = 0x4b58,
+ },
#endif
/* Alder Lake */
--
cgit
From c4824ae7db418aee6f50f308a20b832e58e997fd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jul 2021 11:26:40 +0200
Subject: ALSA: pcm: Fix mmap capability check
The hw_support_mmap() doesn't cover all memory allocation types and
might use a wrong device pointer for checking the capability.
Check the all memory allocation types more completely.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210720092640.12338-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 14 ++++++++++----
1 file changed, 10 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index c88c4316c417..6919d2943b9d 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -246,12 +246,18 @@ 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 ||
- (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV &&
- substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC))
+ if (substream->ops->mmap)
return true;
- return dma_can_mmap(substream->dma_buffer.dev.dev);
+ switch (substream->dma_buffer.dev.type) {
+ case SNDRV_DMA_TYPE_UNKNOWN:
+ return false;
+ case SNDRV_DMA_TYPE_CONTINUOUS:
+ case SNDRV_DMA_TYPE_VMALLOC:
+ return true;
+ default:
+ return dma_can_mmap(substream->dma_buffer.dev.dev);
+ }
}
static int constrain_mask_params(struct snd_pcm_substream *substream,
--
cgit
From d371588910715ebf7fa8e3a5d21ea5169c852927 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Jul 2021 11:27:32 +0200
Subject: ALSA: pcm: Fix mmap without buffer preallocation
The recent rewrite of the memory allocation helpers also changed the
page extraction to a common helper, snd_sgbuf_get_page(). But this
assumes implicitly that the buffer was allocated via the standard
helper (usually via preallocation), and didn't consider the case of
the manual buffer handling.
This patch fixes it and also covers the manual buffer management.
Fixes: 37af81c5998f ("ALSA: core: Abstract memory alloc helpers")
Link: https://lore.kernel.org/r/20210720092732.12412-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 6919d2943b9d..6a2971a7e6a1 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -3675,6 +3675,8 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf)
return VM_FAULT_SIGBUS;
if (substream->ops->page)
page = substream->ops->page(substream, offset);
+ else if (!snd_pcm_get_dma_buf(substream))
+ page = virt_to_page(runtime->dma_area + offset);
else
page = snd_sgbuf_get_page(snd_pcm_get_dma_buf(substream), offset);
if (!page)
--
cgit
From e81d71e343c6c62cf323042caed4b7ca049deda5 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Tue, 20 Jul 2021 18:32:16 +0300
Subject: ALSA: hda/hdmi: Add quirk to force pin connectivity on NUC10
On some Intel NUC10 variants, codec reports AC_JACK_PORT_NONE as
pin default config for all pins. This results in broken audio.
Add a quirk to force connectivity.
BugLink: https://github.com/clearlinux/distribution/issues/2396
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210720153216.2200938-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_hdmi.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 84c088912b3c..e143e69d8184 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1941,6 +1941,7 @@ static const struct snd_pci_quirk force_connect_list[] = {
SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
+ SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1),
{}
};
--
cgit
From b0084afde27fe8a504377dee65f55bc6aa776937 Mon Sep 17 00:00:00 2001
From: Alexander Tsoy <alexander@tsoy.me>
Date: Thu, 22 Jul 2021 02:56:05 +0300
Subject: ALSA: usb-audio: Add registration quirk for JBL Quantum headsets
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
These devices has two interfaces, but only the second interface
contains the capture endpoint, thus quirk is required to delay the
registration until the second interface appears.
Tested-by: Jakub FiĊĦer <jakub@ufiseru.cz>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210721235605.53741-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/quirks.c | 3 +++
1 file changed, 3 insertions(+)
(limited to 'sound')
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 8b8bee3c3dd6..e7accd87e063 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1897,6 +1897,9 @@ static const struct registration_quirk registration_quirks[] = {
REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */
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, 0x203e, 2), /* JBL Quantum 800 */
{ 0 } /* terminator */
};
--
cgit
From cdf72837cda89b2d38bd18fbe6cc591c1d5f2416 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
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 <g@b4.vu>
Link: https://lore.kernel.org/r/77f1000652c37e3217fb8dad8e156bc6392abc0b.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From d3a4f784d20c696b134b916f57956f12a37ecd47 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
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 <g@b4.vu>
Link: https://lore.kernel.org/r/faf5de1d2100038e7d07520d770fda4a1adc276a.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 9ee0fc8366ddce380547878640708f1bd7dd2ead Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
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 <g@b4.vu>
Link: https://lore.kernel.org/r/aa18ddbf8d8bd7f31832ab1b6b6057c00b931202.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 2b8b12be9b9752c36efda38b7dd5d83d790d01d8 Mon Sep 17 00:00:00 2001
From: "Geoffrey D. Bennett" <g@b4.vu>
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 <g@b4.vu>
Link: https://lore.kernel.org/r/8192e15ba62fa4bc90425c005f265c0de530be20.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 4511781f95da0a3b2bad34f3f5e3967e80cd2d18 Mon Sep 17 00:00:00 2001
From: "chihhao.chen" <chihhao.chen@mediatek.com>
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 <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/1627100621-19225-1-git-send-email-chihhao.chen@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 66291b6adb66dd3bc96b0f594d88c2ff1300d95f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
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: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hr1flh9ov.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 35171fbfc0d94aa31b009bb475d156ad1941ab50 Mon Sep 17 00:00:00 2001
From: Nikos Liolios <liolios.nk@gmail.com>
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 <liolios.nk@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727030510.36292-1-liolios.nk@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 4b0556b96e1fe7723629bd40e3813a30cd632faf Mon Sep 17 00:00:00 2001
From: Alexander Tsoy <alexander@tsoy.me>
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 <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727093326.1153366-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 0d4867a185460397af56b9afe3e2243d3e610e37 Mon Sep 17 00:00:00 2001
From: Alexander Monakov <amonakov@ispras.ru>
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 <amonakov@ispras.ru>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210721170141.24807-1-amonakov@ispras.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From 8dde723fcde4479f256441da03793e37181d9f21 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
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: <stable@vger.kernel.org>
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 <tiwai@suse.de>
---
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
From 852a8a97776a153be2e6c803218eced45f37a19c Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
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: <stable@vger.kernel.org>
Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210730090254.612478-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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
From eda80d7c9c4db0f55f130e38c682e19b58d5add7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
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 <hdegoede@redhat.com>
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 <tiwai@suse.de>
---
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
From 97367c97226aab8b298ada954ce12659ee3ad2a4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
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 <folkert@vanheusden.com>
Cc: <stable@vger.kernel.org>
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 <tiwai@suse.de>
---
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
From 739d0959fbed23838a96c48fbce01dd2f6fb2c5f Mon Sep 17 00:00:00 2001
From: Luke D Jones <luke@ljones.dev>
Date: Sat, 7 Aug 2021 14:58:05 +1200
Subject: ALSA: hda: Add quirk for ASUS Flow x13
The ASUS GV301QH sound appears to work well with the quirk for
ALC294_FIXUP_ASUS_DUAL_SPK.
Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210807025805.27321-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 21c521596c9d..4043a2362f27 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8465,6 +8465,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS),
SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
+ SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK),
SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
--
cgit
From dc0dc8a73e8e4dc33fba93dfe23356cc5a500c57 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 8 Aug 2021 09:01:16 +0200
Subject: ALSA: pcm: Fix mmap breakage without explicit buffer setup
The recent fix c4824ae7db41 ("ALSA: pcm: Fix mmap capability check")
restricts the mmap capability only to the drivers that properly set up
the buffers, but it caused a regression for a few drivers that manage
the buffer on its own way.
For those with UNKNOWN buffer type (i.e. the uninitialized / unused
substream->dma_buffer), just assume that the driver handles the mmap
properly and blindly trust the hardware info bit.
Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check")
Reported-and-tested-by: Jeff Woods <jwoods@fnordco.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5him0gpghv.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/core/pcm_native.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 09c0e2a6489c..71323d807dbf 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -251,7 +251,10 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream)
switch (substream->dma_buffer.dev.type) {
case SNDRV_DMA_TYPE_UNKNOWN:
- return false;
+ /* we can't know the device, so just assume that the driver does
+ * everything right
+ */
+ return true;
case SNDRV_DMA_TYPE_CONTINUOUS:
case SNDRV_DMA_TYPE_VMALLOC:
return true;
--
cgit
From d07149aba2ef423eae94a9cc2a6365d0cdf6fd51 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Tue, 10 Aug 2021 18:08:45 +0800
Subject: ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 650 G8
Notebook PC
The HP ProBook 650 G8 Notebook PC is using ALC236 codec which is
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 <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210810100846.65844-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 4043a2362f27..a065260d0d20 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8431,6 +8431,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
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, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED),
--
cgit
From a2befe9380dd04ee76c871568deca00eedf89134 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 11 Aug 2021 18:14:41 +0200
Subject: ALSA: hda - fix the 'Capture Switch' value change notifications
The original code in the cap_put_caller() function does not
handle correctly the positive values returned from the passed
function for multiple iterations. It means that the change
notifications may be lost.
Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213851
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210811161441.1325250-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_generic.c | 10 +++++++---
1 file changed, 7 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e97d00585e8e..481d8f8d3396 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3460,7 +3460,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol,
struct hda_gen_spec *spec = codec->spec;
const struct hda_input_mux *imux;
struct nid_path *path;
- int i, adc_idx, err = 0;
+ int i, adc_idx, ret, err = 0;
imux = &spec->input_mux;
adc_idx = kcontrol->id.index;
@@ -3470,9 +3470,13 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol,
if (!path || !path->ctls[type])
continue;
kcontrol->private_value = path->ctls[type];
- err = func(kcontrol, ucontrol);
- if (err < 0)
+ ret = func(kcontrol, ucontrol);
+ if (ret < 0) {
+ err = ret;
break;
+ }
+ if (ret > 0)
+ err = 1;
}
mutex_unlock(&codec->control_mutex);
if (err >= 0 && spec->cap_sync_hook)
--
cgit
From 67bb66d32905627e29400e2cb7f87a7c4c8cf667 Mon Sep 17 00:00:00 2001
From: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Date: Thu, 12 Aug 2021 11:28:39 +0900
Subject: ALSA: oxfw: fix functioal regression for silence in Apogee Duet
FireWire
OXFW 971 has no function to use the value in syt field of received
isochronous packet for playback timing generation. In kernel prepatch for
v5.14, ALSA OXFW driver got change to send NO_INFO value in the field
instead of actual timing value. The change brings Apogee Duet FireWire to
generate no playback sound, while output meter moves.
As long as I investigate, _any_ value in the syt field takes the device to
generate sound. It's reasonable to think that the device just ignores data
blocks in packet with NO_INFO value in its syt field for audio data
processing.
This commit adds a new flag for the quirk to fix regression.
Fixes: 029ffc429440 ("ALSA: oxfw: perform sequence replay for media clock recovery")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210812022839.42043-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/firewire/oxfw/oxfw-stream.c | 9 ++++++++-
sound/firewire/oxfw/oxfw.c | 6 ++++--
sound/firewire/oxfw/oxfw.h | 5 +++++
3 files changed, 17 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 0ef242fdd3bc..fff18b5d4e05 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -153,7 +153,7 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream)
struct cmp_connection *conn;
enum cmp_direction c_dir;
enum amdtp_stream_direction s_dir;
- unsigned int flags = CIP_UNAWARE_SYT;
+ unsigned int flags = 0;
int err;
if (!(oxfw->quirks & SND_OXFW_QUIRK_BLOCKING_TRANSMISSION))
@@ -161,6 +161,13 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream)
else
flags |= CIP_BLOCKING;
+ // OXFW 970/971 has no function to generate playback timing according to the sequence
+ // of value in syt field, thus the packet should include NO_INFO value in the field.
+ // However, some models just ignore data blocks in packet with NO_INFO for audio data
+ // processing.
+ if (!(oxfw->quirks & SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET))
+ flags |= CIP_UNAWARE_SYT;
+
if (stream == &oxfw->tx_stream) {
conn = &oxfw->out_conn;
c_dir = CMP_OUTPUT;
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 84971d78d152..cb5b5e3a481b 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -159,8 +159,10 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id
return snd_oxfw_scs1x_add(oxfw);
}
- if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW)
- oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION;
+ if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW) {
+ oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION |
+ SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET;
+ }
/*
* TASCAM FireOne has physical control and requires a pair of additional
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index ee47abcb0c90..c13034f6c2ca 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -42,6 +42,11 @@ enum snd_oxfw_quirk {
SND_OXFW_QUIRK_BLOCKING_TRANSMISSION = 0x04,
// Stanton SCS1.d and SCS1.m support unique transaction.
SND_OXFW_QUIRK_SCS_TRANSACTION = 0x08,
+ // Apogee Duet FireWire ignores data blocks in packet with NO_INFO for audio data
+ // processing, while output level meter moves. Any value in syt field of packet takes
+ // the device to process audio data even if the value is invalid in a point of
+ // IEC 61883-1/6.
+ SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET = 0x10,
};
/* This is an arbitrary number for convinience. */
--
cgit
From da94692001ea45ffa1f5e9f17ecdef7aecd90c27 Mon Sep 17 00:00:00 2001
From: Kristin Paget <kristin@tombom.co.uk>
Date: Sat, 14 Aug 2021 15:46:05 -0700
Subject: ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9510
laptop
The 2021-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio
setup as the Precision models, so requires the same quirk to enable woofer
output. Tested on my own 9510.
Signed-off-by: Kristin Paget <kristin@tombom.co.uk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/e1fc95c5-c10a-1f98-a5c2-dd6e336157e1@tombom.co.uk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 a065260d0d20..96f32eaa24df 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8332,6 +8332,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
--
cgit
From 0165c4e19f6ec76b535de090e4bd145c73810c51 Mon Sep 17 00:00:00 2001
From: Imre Deak <imre.deak@intel.com>
Date: Mon, 16 Aug 2021 20:42:59 +0300
Subject: ALSA: hda: Fix hang during shutdown due to link reset
During system shutdown codecs may be still active, and resetting the
controller->codec HW link in this state - based on the bug reporter's
tests - leads to the shutdown sequence to get stuck. This happens at
least on the reporter's KBL system with an ALC662 codec.
For now fix the issue by skipping the link reset step.
Fixes: 472e18f63c42 ("ALSA: hda: Release controller display power during shutdown/reboot")
References: https://bugzilla.kernel.org/show_bug.cgi?id=214045
References: https://gitlab.freedesktop.org/drm/intel/-/issues/3618#note_1024665
Reported-and-tested-by: youling257@gmail.com
Cc: youling257@gmail.com
Signed-off-by: Imre Deak <imre.deak@intel.com>
Link: https://lore.kernel.org/r/20210816174259.2759103-1-imre.deak@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 12 +++++++++---
1 file changed, 9 insertions(+), 3 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0322b289505e..0062c18b646a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev)
return azx_get_pos_posbuf(chip, azx_dev);
}
-static void azx_shutdown_chip(struct azx *chip)
+static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset)
{
azx_stop_chip(chip);
- azx_enter_link_reset(chip);
+ if (!skip_link_reset)
+ azx_enter_link_reset(chip);
azx_clear_irq_pending(chip);
display_power(chip, false);
}
@@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip)
static DEFINE_MUTEX(card_list_lock);
static LIST_HEAD(card_list);
+static void azx_shutdown_chip(struct azx *chip)
+{
+ __azx_shutdown_chip(chip, false);
+}
+
static void azx_add_card_list(struct azx *chip)
{
struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
@@ -2385,7 +2391,7 @@ static void azx_shutdown(struct pci_dev *pci)
return;
chip = card->private_data;
if (chip && chip->running)
- azx_shutdown_chip(chip);
+ __azx_shutdown_chip(chip, true);
}
/* PCI IDs */
--
cgit
From 4bf61ad5f0204b67ba570da6e5c052c2095e29df Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Aug 2021 07:24:32 +0200
Subject: ALSA: hda/via: Apply runtime PM workaround for ASUS B23E
ASUS B23E requires the same workaround like other machines with
VT1802, otherwise it looses the codec power on a few nodes and the
sound kept silence.
Fixes: a0645daf1610 ("ALSA: HDA: Early Forbid of runtime PM")
Link: https://lore.kernel.org/r/ac2232f142efcd67fe6ac38897f704f7176bd200.camel@gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210817052432.14751-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_via.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a5c1a2c4eae4..773a136161f1 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1041,6 +1041,7 @@ static const struct hda_fixup via_fixups[] = {
};
static const struct snd_pci_quirk vt2002p_fixups[] = {
+ SND_PCI_QUIRK(0x1043, 0x13f7, "Asus B23E", VIA_FIXUP_POWER_SAVE),
SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75),
SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE),
--
cgit
From 8903376dc69949199301b290cc22dc64ae5d8a6d Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Wed, 18 Aug 2021 22:41:18 +0800
Subject: ALSA: hda/realtek: Limit mic boost on HP ProBook 445 G8
The mic has lots of noises if mic boost is enabled. So disable mic boost
to get crystal clear audio capture.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210818144119.121738-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 11 +++++++++--
1 file changed, 9 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 96f32eaa24df..7ad689f991e7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6658,6 +6658,7 @@ enum {
ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP,
ALC623_FIXUP_LENOVO_THINKSTATION_P340,
ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
+ ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC
},
+ [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ .chained = true,
+ .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8438,8 +8445,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 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, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
- SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+ SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST),
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),
--
cgit
From 65ca89c2b12cca0d473f3dd54267568ad3af55cc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Aug 2021 17:29:45 +0200
Subject: ASoC: intel: atom: Fix breakage for PCM buffer address setup
The commit 2e6b836312a4 ("ASoC: intel: atom: Fix reference to PCM
buffer address") changed the reference of PCM buffer address to
substream->runtime->dma_addr as the buffer address may change
dynamically. However, I forgot that the dma_addr field is still not
set up for the CONTINUOUS buffer type (that this driver uses) yet in
5.14 and earlier kernels, and it resulted in garbage I/O. The problem
will be fixed in 5.15, but we need to address it quickly for now.
The fix is to deduce the address again from the DMA pointer with
virt_to_phys(), but from the right one, substream->runtime->dma_area.
Fixes: 2e6b836312a4 ("ASoC: intel: atom: Fix reference to PCM buffer address")
Reported-and-tested-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@vger.kernel.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/2048c6aa-2187-46bd-6772-36a4fb3c5aeb@redhat.com
Link: https://lore.kernel.org/r/20210819152945.8510-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 5db2f4865bbb..905c7965f653 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream,
snd_pcm_uframes_t period_size;
ssize_t periodbytes;
ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
- u32 buffer_addr = substream->runtime->dma_addr;
+ u32 buffer_addr = virt_to_phys(substream->runtime->dma_area);
channels = substream->runtime->channels;
period_size = substream->runtime->period_size;
--
cgit