From 1ef8715975de8bd481abbd0839ed4f49d9e5b0ff Mon Sep 17 00:00:00 2001
From: Borislav Petkov <bp@suse.de>
Date: Tue, 5 Apr 2022 17:15:08 +0200
Subject: ALSA: usb-audio: Fix undefined behavior due to shift overflowing the
 constant
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fix:

  sound/usb/midi.c: In function ‘snd_usbmidi_out_endpoint_create’:
  sound/usb/midi.c:1389:2: error: case label does not reduce to an integer constant
    case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */
    ^~~~

See https://lore.kernel.org/r/YkwQ6%2BtIH8GQpuct@zn.tnic for the gory
details as to why it triggers with older gccs only.

[ A slight correction with parentheses around the argument by tiwai ]

Signed-off-by: Borislav Petkov <bp@suse.de>
Link: https://lore.kernel.org/r/20220405151517.29753-3-bp@alien8.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbaudio.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/usb')

diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 167834133b9b..b8359a0aa008 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -8,7 +8,7 @@
  */
 
 /* handling of USB vendor/product ID pairs as 32-bit numbers */
-#define USB_ID(vendor, product) (((vendor) << 16) | (product))
+#define USB_ID(vendor, product) (((unsigned int)(vendor) << 16) | (product))
 #define USB_ID_VENDOR(id) ((id) >> 16)
 #define USB_ID_PRODUCT(id) ((u16)(id))
 
-- 
cgit v1.2.3


From 98c27add5d96485db731a92dac31567b0486cae8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 7 Apr 2022 23:16:57 +0200
Subject: ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit
 fb

In the implicit feedback mode, some parameters are tied between both
playback and capture streams.  One of the tied parameters is the
period size, and this can be a problem if the device has different
number of channels to both streams.  Assume that an application opens
a playback stream that has an implicit feedback from a capture stream,
and it allocates up to the max period and buffer size as much as
possible.  When the capture device supports only more channels than
the playback, the minimum period and buffer sizes become larger than
the sizes the playback stream took.  That is, the minimum size will be
over the max size the driver limits, and PCM core sees as if no
available configuration is found, returning -EINVAL mercilessly.

For avoiding this problem, we have to look through the counter part of
audioformat list for each sync ep, and checks the channels.  If more
channels are found there, we reduce the max period and buffer sizes
accordingly.

You may wonder that the patch adds only the evaluation of channels
between streams, and what about other parameters?  Both the format and
the rate are tied in the implicit fb mode, hence they are always
identical.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215792
Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220407211657.15087-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 89 +++++++++++++++++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 87 insertions(+), 2 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cec6e91afea2..6a460225f2e3 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -659,6 +659,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 #define hwc_debug(fmt, args...) do { } while(0)
 #endif
 
+#define MAX_BUFFER_BYTES	(1024 * 1024)
+#define MAX_PERIOD_BYTES	(512 * 1024)
+
 static const struct snd_pcm_hardware snd_usb_hardware =
 {
 	.info =			SNDRV_PCM_INFO_MMAP |
@@ -669,9 +672,9 @@ static const struct snd_pcm_hardware snd_usb_hardware =
 				SNDRV_PCM_INFO_PAUSE,
 	.channels_min =		1,
 	.channels_max =		256,
-	.buffer_bytes_max =	1024 * 1024,
+	.buffer_bytes_max =	MAX_BUFFER_BYTES,
 	.period_bytes_min =	64,
-	.period_bytes_max =	512 * 1024,
+	.period_bytes_max =	MAX_PERIOD_BYTES,
 	.periods_min =		2,
 	.periods_max =		1024,
 };
@@ -971,6 +974,78 @@ static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params,
 				      ep->cur_buffer_periods);
 }
 
+/* get the adjusted max buffer (or period) bytes that can fit with the
+ * paired format for implicit fb
+ */
+static unsigned int
+get_adjusted_max_bytes(struct snd_usb_substream *subs,
+		       struct snd_usb_substream *pair,
+		       struct snd_pcm_hw_params *params,
+		       unsigned int max_bytes,
+		       bool reverse_map)
+{
+	const struct audioformat *fp, *pp;
+	unsigned int rmax = 0, r;
+
+	list_for_each_entry(fp, &subs->fmt_list, list) {
+		if (!fp->implicit_fb)
+			continue;
+		if (!reverse_map &&
+		    !hw_check_valid_format(subs, params, fp))
+			continue;
+		list_for_each_entry(pp, &pair->fmt_list, list) {
+			if (pp->iface != fp->sync_iface ||
+			    pp->altsetting != fp->sync_altsetting ||
+			    pp->ep_idx != fp->sync_ep_idx)
+				continue;
+			if (reverse_map &&
+			    !hw_check_valid_format(pair, params, pp))
+				break;
+			if (!reverse_map && pp->channels > fp->channels)
+				r = max_bytes * fp->channels / pp->channels;
+			else if (reverse_map && pp->channels < fp->channels)
+				r = max_bytes * pp->channels / fp->channels;
+			else
+				r = max_bytes;
+			rmax = max(rmax, r);
+			break;
+		}
+	}
+	return rmax;
+}
+
+/* Reduce the period or buffer bytes depending on the paired substream;
+ * when a paired configuration for implicit fb has a higher number of channels,
+ * we need to reduce the max size accordingly, otherwise it may become unusable
+ */
+static int hw_rule_bytes_implicit_fb(struct snd_pcm_hw_params *params,
+				     struct snd_pcm_hw_rule *rule)
+{
+	struct snd_usb_substream *subs = rule->private;
+	struct snd_usb_substream *pair;
+	struct snd_interval *it;
+	unsigned int max_bytes;
+	unsigned int rmax;
+
+	pair = &subs->stream->substream[!subs->direction];
+	if (!pair->ep_num)
+		return 0;
+
+	if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_BYTES)
+		max_bytes = MAX_PERIOD_BYTES;
+	else
+		max_bytes = MAX_BUFFER_BYTES;
+
+	rmax = get_adjusted_max_bytes(subs, pair, params, max_bytes, false);
+	if (!rmax)
+		rmax = get_adjusted_max_bytes(pair, subs, params, max_bytes, true);
+	if (!rmax)
+		return 0;
+
+	it = hw_param_interval(params, rule->var);
+	return apply_hw_params_minmax(it, 0, rmax);
+}
+
 /*
  * set up the runtime hardware information.
  */
@@ -1085,6 +1160,16 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
 				  SNDRV_PCM_HW_PARAM_PERIODS, -1);
 	if (err < 0)
 		return err;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+				  hw_rule_bytes_implicit_fb, subs,
+				  SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1);
+	if (err < 0)
+		return err;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+				  hw_rule_bytes_implicit_fb, subs,
+				  SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1);
+	if (err < 0)
+		return err;
 
 	list_for_each_entry(fp, &subs->fmt_list, list) {
 		if (fp->implicit_fb) {
-- 
cgit v1.2.3


From fee2ec8cceb33b8886bc5894fb07e0b2e34148af Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 7 Apr 2022 23:27:40 +0200
Subject: ALSA: usb-audio: Increase max buffer size

The current limit of max buffer size 1MB seems too small for modern
devices with lots of channels and high sample rates.
Let's make bigger, 4MB.

Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220407212740.17920-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/usb')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 6a460225f2e3..37ee6df8b15a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -659,7 +659,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 #define hwc_debug(fmt, args...) do { } while(0)
 #endif
 
-#define MAX_BUFFER_BYTES	(1024 * 1024)
+#define MAX_BUFFER_BYTES	(4 * 1024 * 1024)
 #define MAX_PERIOD_BYTES	(512 * 1024)
 
 static const struct snd_pcm_hardware snd_usb_hardware =
-- 
cgit v1.2.3


From 24d0c9f0e7de95fe3e3e0067cbea1cd5d413244b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 12 Apr 2022 15:07:40 +0200
Subject: ALSA: usb-audio: Limit max buffer and period sizes per time

In the previous fix, we increased the max buffer bytes from 1MB to 4MB
so that we can use bigger buffers for the modern HiFi devices with
higher rates, more channels and wider formats.  OTOH, extending this
has a concern that too big buffer is allowed for the lower rates, less
channels and narrower formats; when an application tries to allocate
as big buffer as possible, it'll lead to unexpectedly too huge size.

Also, we had a problem about the inconsistent max buffer and period
bytes for the implicit feedback mode when both streams have different
channels.  This was fixed by the (relatively complex) patch to reduce
the max buffer and period bytes accordingly.

This is an alternative fix for those, a patch to kill two birds with
one stone (*): instead of increasing the max buffer bytes blindly and
applying the reduction per channels, we simply use the hw constraints
for the buffer and period "time".  Meanwhile the max buffer and period
bytes are set unlimited instead.

Since the inconsistency of buffer (and period) bytes comes from the
difference of the channels in the tied streams, as long as we care
only about the buffer (and period) time, it doesn't matter; the buffer
time is same for different channels, although we still allow higher
buffer size.  Similarly, this will allow more buffer bytes for HiFi
devices while it also keeps the reasonable size for the legacy
devices, too.

As of this patch, the max period and buffer time are set to 1 and 2
seconds, which should be large enough for all possible use cases.

(*) No animals were harmed in the making of this patch.

Fixes: 98c27add5d96 ("ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit fb")
Fixes: fee2ec8cceb3 ("ALSA: usb-audio: Increase max buffer size")
Link: https://lore.kernel.org/r/20220412130740.18933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 101 ++++++++------------------------------------------------
 1 file changed, 14 insertions(+), 87 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 37ee6df8b15a..6d699065e81a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -659,9 +659,6 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 #define hwc_debug(fmt, args...) do { } while(0)
 #endif
 
-#define MAX_BUFFER_BYTES	(4 * 1024 * 1024)
-#define MAX_PERIOD_BYTES	(512 * 1024)
-
 static const struct snd_pcm_hardware snd_usb_hardware =
 {
 	.info =			SNDRV_PCM_INFO_MMAP |
@@ -672,9 +669,9 @@ static const struct snd_pcm_hardware snd_usb_hardware =
 				SNDRV_PCM_INFO_PAUSE,
 	.channels_min =		1,
 	.channels_max =		256,
-	.buffer_bytes_max =	MAX_BUFFER_BYTES,
+	.buffer_bytes_max =	INT_MAX, /* limited by BUFFER_TIME later */
 	.period_bytes_min =	64,
-	.period_bytes_max =	MAX_PERIOD_BYTES,
+	.period_bytes_max =	INT_MAX, /* limited by PERIOD_TIME later */
 	.periods_min =		2,
 	.periods_max =		1024,
 };
@@ -974,78 +971,6 @@ static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params,
 				      ep->cur_buffer_periods);
 }
 
-/* get the adjusted max buffer (or period) bytes that can fit with the
- * paired format for implicit fb
- */
-static unsigned int
-get_adjusted_max_bytes(struct snd_usb_substream *subs,
-		       struct snd_usb_substream *pair,
-		       struct snd_pcm_hw_params *params,
-		       unsigned int max_bytes,
-		       bool reverse_map)
-{
-	const struct audioformat *fp, *pp;
-	unsigned int rmax = 0, r;
-
-	list_for_each_entry(fp, &subs->fmt_list, list) {
-		if (!fp->implicit_fb)
-			continue;
-		if (!reverse_map &&
-		    !hw_check_valid_format(subs, params, fp))
-			continue;
-		list_for_each_entry(pp, &pair->fmt_list, list) {
-			if (pp->iface != fp->sync_iface ||
-			    pp->altsetting != fp->sync_altsetting ||
-			    pp->ep_idx != fp->sync_ep_idx)
-				continue;
-			if (reverse_map &&
-			    !hw_check_valid_format(pair, params, pp))
-				break;
-			if (!reverse_map && pp->channels > fp->channels)
-				r = max_bytes * fp->channels / pp->channels;
-			else if (reverse_map && pp->channels < fp->channels)
-				r = max_bytes * pp->channels / fp->channels;
-			else
-				r = max_bytes;
-			rmax = max(rmax, r);
-			break;
-		}
-	}
-	return rmax;
-}
-
-/* Reduce the period or buffer bytes depending on the paired substream;
- * when a paired configuration for implicit fb has a higher number of channels,
- * we need to reduce the max size accordingly, otherwise it may become unusable
- */
-static int hw_rule_bytes_implicit_fb(struct snd_pcm_hw_params *params,
-				     struct snd_pcm_hw_rule *rule)
-{
-	struct snd_usb_substream *subs = rule->private;
-	struct snd_usb_substream *pair;
-	struct snd_interval *it;
-	unsigned int max_bytes;
-	unsigned int rmax;
-
-	pair = &subs->stream->substream[!subs->direction];
-	if (!pair->ep_num)
-		return 0;
-
-	if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_BYTES)
-		max_bytes = MAX_PERIOD_BYTES;
-	else
-		max_bytes = MAX_BUFFER_BYTES;
-
-	rmax = get_adjusted_max_bytes(subs, pair, params, max_bytes, false);
-	if (!rmax)
-		rmax = get_adjusted_max_bytes(pair, subs, params, max_bytes, true);
-	if (!rmax)
-		return 0;
-
-	it = hw_param_interval(params, rule->var);
-	return apply_hw_params_minmax(it, 0, rmax);
-}
-
 /*
  * set up the runtime hardware information.
  */
@@ -1139,6 +1064,18 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
 			return err;
 	}
 
+	/* set max period and buffer sizes for 1 and 2 seconds, respectively */
+	err = snd_pcm_hw_constraint_minmax(runtime,
+					   SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+					   0, 1000000);
+	if (err < 0)
+		return err;
+	err = snd_pcm_hw_constraint_minmax(runtime,
+					   SNDRV_PCM_HW_PARAM_BUFFER_TIME,
+					   0, 2000000);
+	if (err < 0)
+		return err;
+
 	/* additional hw constraints for implicit fb */
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
 				  hw_rule_format_implicit_fb, subs,
@@ -1160,16 +1097,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
 				  SNDRV_PCM_HW_PARAM_PERIODS, -1);
 	if (err < 0)
 		return err;
-	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
-				  hw_rule_bytes_implicit_fb, subs,
-				  SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1);
-	if (err < 0)
-		return err;
-	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
-				  hw_rule_bytes_implicit_fb, subs,
-				  SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1);
-	if (err < 0)
-		return err;
 
 	list_for_each_entry(fp, &subs->fmt_list, list) {
 		if (fp->implicit_fb) {
-- 
cgit v1.2.3


From 4ddef9c4d70aae0c9029bdec7c3f7f1c1c51ff8c Mon Sep 17 00:00:00 2001
From: Maurizio Avogadro <mavoga@gmail.com>
Date: Mon, 18 Apr 2022 15:16:12 +0200
Subject: ALSA: usb-audio: add mapping for MSI MAG X570S Torpedo MAX.

The USB audio device 0db0:a073 based on the Realtek ALC4080 chipset
exposes all playback volume controls as "PCM". This makes
distinguishing the individual functions hard.
The mapping already adopted for device 0db0:419c based on the same
chipset fixes the issue, apply it for this device too.

Signed-off-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/Yl1ykPaGgsFf3SnW@ryzen
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_maps.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 64f5544d0a0a..7ef7a8abcc2b 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -599,6 +599,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x0db0, 0x419c),
 		.map = msi_mpg_x570s_carbon_max_wifi_alc4080_map,
 	},
+	{	/* MSI MAG X570S Torpedo Max */
+		.id = USB_ID(0x0db0, 0xa073),
+		.map = msi_mpg_x570s_carbon_max_wifi_alc4080_map,
+	},
 	{	/* MSI TRX40 */
 		.id = USB_ID(0x0db0, 0x543d),
 		.map = trx40_mobo_map,
-- 
cgit v1.2.3


From 0665886ad1392e6b5bae85d7a6ccbed48dca1522 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Apr 2022 15:02:47 +0200
Subject: ALSA: usb-audio: Clear MIDI port active flag after draining

When a rawmidi output stream is closed, it calls the drain at first,
then does trigger-off only when the drain returns -ERESTARTSYS as a
fallback.  It implies that each driver should turn off the stream
properly after the drain.  Meanwhile, USB-audio MIDI interface didn't
change the port->active flag after the drain.  This may leave the
output work picking up the port that is closed right now, which
eventually leads to a use-after-free for the already released rawmidi
object.

This patch fixes the bug by properly clearing the port->active flag
after the output drain.

Reported-by: syzbot+70e777a39907d6d5fd0a@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/00000000000011555605dceaff03@google.com
Link: https://lore.kernel.org/r/20220420130247.22062-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/midi.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/usb')

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 2c01649c70f6..7c6ca2b433a5 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1194,6 +1194,7 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
 		} while (drain_urbs && timeout);
 		finish_wait(&ep->drain_wait, &wait);
 	}
+	port->active = 0;
 	spin_unlock_irq(&ep->buffer_lock);
 }
 
-- 
cgit v1.2.3


From 0f1f7a6661394fe4a53db254c346d6aa2dd64397 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Apr 2022 08:41:00 +0200
Subject: ALSA: usb-audio: Add quirk bits for enabling/disabling generic
 implicit fb

For making easier to test, add the new quirk_flags bits 17 and 18 to
enable and disable the generic implicit feedback mode.  The bit 17 is
equivalent with implicit_fb=1 option, applying the generic implicit
feedback sync mode.  OTOH, the bit 18 disables the implicit fb mode
forcibly.

Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa-configuration.rst | 4 +++-
 sound/usb/implicit.c                       | 5 ++++-
 sound/usb/usbaudio.h                       | 6 ++++++
 3 files changed, 13 insertions(+), 2 deletions(-)

(limited to 'sound/usb')

diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index 34888d4fc4a8..21ab5e6f7062 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -2246,7 +2246,7 @@ implicit_fb
     Apply the generic implicit feedback sync mode.  When this is set
     and the playback stream sync mode is ASYNC, the driver tries to
     tie an adjacent ASYNC capture stream as the implicit feedback
-    source.
+    source.  This is equivalent with quirk_flags bit 17.
 use_vmalloc
     Use vmalloc() for allocations of the PCM buffers (default: yes).
     For architectures with non-coherent memory like ARM or MIPS, the
@@ -2288,6 +2288,8 @@ quirk_flags
         * bit 14: Ignore errors for mixer access
         * bit 15: Support generic DSD raw U32_BE format
         * bit 16: Set up the interface at first like UAC1
+        * bit 17: Apply the generic implicit feedback sync mode
+        * bit 18: Don't apply implicit feedback sync mode
 
 This module supports multiple devices, autoprobe and hotplugging.
 
diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 2d444ec74202..1fd087128538 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip,
 	}
 
 	/* Try the generic implicit fb if available */
-	if (chip->generic_implicit_fb)
+	if (chip->generic_implicit_fb ||
+	    (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB))
 		return add_generic_implicit_fb(chip, fmt, alts);
 
 	/* No quirk */
@@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip,
 				    struct audioformat *fmt,
 				    struct usb_host_interface *alts)
 {
+	if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB)
+		return 0;
 	if (fmt->endpoint & USB_DIR_IN)
 		return audioformat_capture_quirk(chip, fmt, alts);
 	else
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b8359a0aa008..044cd7ab27cb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation;
  *  Support generic DSD raw U32_BE format
  * QUIRK_FLAG_SET_IFACE_FIRST:
  *  Set up the interface at first like UAC1
+ * QUIRK_FLAG_GENERIC_IMPLICIT_FB
+ *  Apply the generic implicit feedback sync mode (same as implicit_fb=1 option)
+ * QUIRK_FLAG_SKIP_IMPLICIT_FB
+ *  Don't apply implicit feedback sync mode
  */
 
 #define QUIRK_FLAG_GET_SAMPLE_RATE	(1U << 0)
@@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation;
 #define QUIRK_FLAG_IGNORE_CTL_ERROR	(1U << 14)
 #define QUIRK_FLAG_DSD_RAW		(1U << 15)
 #define QUIRK_FLAG_SET_IFACE_FIRST	(1U << 16)
+#define QUIRK_FLAG_GENERIC_IMPLICIT_FB	(1U << 17)
+#define QUIRK_FLAG_SKIP_IMPLICIT_FB	(1U << 18)
 
 #endif /* __USBAUDIO_H */
-- 
cgit v1.2.3


From 67d64069bc0867e52e73a1e255b17462005ca9b4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Apr 2022 08:41:01 +0200
Subject: ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c

Use the new quirk bits to manage the generic implicit fb quirk
entries.  This makes easier to compare with other devices.

Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/implicit.c | 5 -----
 sound/usb/quirks.c   | 6 ++++++
 2 files changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c
index 1fd087128538..e1bf1b5da423 100644
--- a/sound/usb/implicit.c
+++ b/sound/usb/implicit.c
@@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match {
 
 /* Implicit feedback quirk table for playback */
 static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = {
-	/* Generic matching */
-	IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */
-	IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */
-	IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */
-
 	/* Fixed EP */
 	/* FIXME: check the availability of generic matching */
 	IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ab9f3da49941..5461cdf907e2 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */
 		   QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR),
+	DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
 	DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */
 		   QUIRK_FLAG_GET_SAMPLE_RATE),
 	DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */
@@ -1824,6 +1826,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */
 		   QUIRK_FLAG_GET_SAMPLE_RATE),
+	DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
+	DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */
+		   QUIRK_FLAG_GENERIC_IMPLICIT_FB),
 	DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */
-- 
cgit v1.2.3


From 80b2b03bacfc071ceaac034fa81d535d27d57b31 Mon Sep 17 00:00:00 2001
From: Vincent Mailhol <mailhol.vincent@wanadoo.fr>
Date: Thu, 17 Mar 2022 12:55:12 +0900
Subject: sound: usb: remove third argument of usb_maxpacket()

The third argument of usb_maxpacket(): in_out has been deprecated
because it could be derived from the second argument (e.g. using
usb_pipeout(pipe)).

N.B. function usb_maxpacket() was made variadic to accommodate the
transition from the old prototype with three arguments to the new one
with only two arguments (so that no renaming is needed). The variadic
argument is to be removed once all users of usb_maxpacket() get
migrated.

CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: Clemens Ladisch <clemens@ladisch.de>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Vincent Mailhol <mailhol.vincent@wanadoo.fr>
Link: https://lore.kernel.org/r/20220317035514.6378-8-mailhol.vincent@wanadoo.fr
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
 sound/usb/line6/pcm.c           | 4 ++--
 sound/usb/midi.c                | 4 ++--
 sound/usb/usx2y/usb_stream.c    | 6 +++---
 sound/usb/usx2y/usbusx2yaudio.c | 2 +-
 sound/usb/usx2y/usx2yhwdeppcm.c | 2 +-
 5 files changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c
index fdbdfb7bce92..6a4af725aedd 100644
--- a/sound/usb/line6/pcm.c
+++ b/sound/usb/line6/pcm.c
@@ -552,10 +552,10 @@ int line6_init_pcm(struct usb_line6 *line6,
 
 	line6pcm->max_packet_size_in =
 		usb_maxpacket(line6->usbdev,
-			usb_rcvisocpipe(line6->usbdev, ep_read), 0);
+			usb_rcvisocpipe(line6->usbdev, ep_read));
 	line6pcm->max_packet_size_out =
 		usb_maxpacket(line6->usbdev,
-			usb_sndisocpipe(line6->usbdev, ep_write), 1);
+			usb_sndisocpipe(line6->usbdev, ep_write));
 	if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) {
 		dev_err(line6pcm->line6->ifcdev,
 			"cannot get proper max packet size\n");
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 2c01649c70f6..fba498f9e7dc 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1285,7 +1285,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi,
 		pipe = usb_rcvintpipe(umidi->dev, ep_info->in_ep);
 	else
 		pipe = usb_rcvbulkpipe(umidi->dev, ep_info->in_ep);
-	length = usb_maxpacket(umidi->dev, pipe, 0);
+	length = usb_maxpacket(umidi->dev, pipe);
 	for (i = 0; i < INPUT_URBS; ++i) {
 		buffer = usb_alloc_coherent(umidi->dev, length, GFP_KERNEL,
 					    &ep->urbs[i]->transfer_dma);
@@ -1374,7 +1374,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
 		pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep);
 	switch (umidi->usb_id) {
 	default:
-		ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1);
+		ep->max_transfer = usb_maxpacket(umidi->dev, pipe);
 		break;
 		/*
 		 * Various chips declare a packet size larger than 4 bytes, but
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 9d0e44793896..a4d32e8a1d36 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -51,7 +51,7 @@ static int init_pipe_urbs(struct usb_stream_kernel *sk,
 {
 	int u, p;
 	int maxpacket = use_packsize ?
-		use_packsize : usb_maxpacket(dev, pipe, usb_pipeout(pipe));
+		use_packsize : usb_maxpacket(dev, pipe);
 	int transfer_length = maxpacket * sk->n_o_ps;
 
 	for (u = 0; u < USB_STREAM_NURBS;
@@ -171,7 +171,7 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
 	out_pipe = usb_sndisocpipe(dev, out_endpoint);
 
 	max_packsize = use_packsize ?
-		use_packsize : usb_maxpacket(dev, in_pipe, 0);
+		use_packsize : usb_maxpacket(dev, in_pipe);
 
 	/*
 		t_period = period_frames / sample_rate
@@ -187,7 +187,7 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
 	read_size += packets * USB_STREAM_URBDEPTH *
 		(max_packsize + sizeof(struct usb_stream_packet));
 
-	max_packsize = usb_maxpacket(dev, out_pipe, 1);
+	max_packsize = usb_maxpacket(dev, out_pipe);
 	write_size = max_packsize * packets * USB_STREAM_URBDEPTH;
 
 	if (read_size >= 256*PAGE_SIZE || write_size >= 256*PAGE_SIZE) {
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index cfc1ea53978d..9cd5e3aae4f7 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -421,7 +421,7 @@ static int usx2y_urbs_allocate(struct snd_usx2y_substream *subs)
 
 	pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) :
 			usb_rcvisocpipe(dev, subs->endpoint);
-	subs->maxpacksize = usb_maxpacket(dev, pipe, is_playback);
+	subs->maxpacksize = usb_maxpacket(dev, pipe);
 	if (!subs->maxpacksize)
 		return -EINVAL;
 
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index db83522c1b49..240349b644f3 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -321,7 +321,7 @@ static int usx2y_usbpcm_urbs_allocate(struct snd_usx2y_substream *subs)
 
 	pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) :
 			usb_rcvisocpipe(dev, subs->endpoint);
-	subs->maxpacksize = usb_maxpacket(dev, pipe, is_playback);
+	subs->maxpacksize = usb_maxpacket(dev, pipe);
 	if (!subs->maxpacksize)
 		return -EINVAL;
 
-- 
cgit v1.2.3


From d7be213849232a2accb219d537edf056d29186b4 Mon Sep 17 00:00:00 2001
From: Forest Crossman <cyrozap@gmail.com>
Date: Tue, 3 May 2022 19:24:44 -0500
Subject: ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI

This device doesn't support reading the sample rate, so we need to apply
this quirk to avoid a 15-second delay waiting for three timeouts.

Signed-off-by: Forest Crossman <cyrozap@gmail.com>
Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ab9f3da49941..fbbe59054c3f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
 	DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */
 		   QUIRK_FLAG_IGNORE_CTL_ERROR),
+	DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */
+		   QUIRK_FLAG_GET_SAMPLE_RATE),
 	DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */
 		   QUIRK_FLAG_GET_SAMPLE_RATE),
 	DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */
-- 
cgit v1.2.3


From 5c62383c06837b5719cd5447a5758b791279e653 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 May 2022 12:31:12 +0200
Subject: ALSA: usb-audio: Restore Rane SL-1 quirk

At cleaning up and moving the device rename from the quirk table to
its own table, we removed the entry for Rane SL-1 as we thought it's
only for renaming.  It turned out, however, that the quirk is required
for matching with the device that declares itself as no standard
audio but only as vendor-specific.

Restore the quirk entry for Rane SL-1 to fix the regression.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215887
Fixes: 5436f59bc5bc ("ALSA: usb-audio: Move device rename and profile quirks to an internal table")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220516103112.12950-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 9 +++++++++
 1 file changed, 9 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 0ea39565e623..40a5e3eb4ef2 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3235,6 +3235,15 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
+/* Rane SL-1 */
+{
+	USB_DEVICE(0x13e5, 0x0001),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_AUDIO_STANDARD_INTERFACE
+        }
+},
+
 /* disabled due to regression for other devices;
  * see https://bugzilla.kernel.org/show_bug.cgi?id=199905
  */
-- 
cgit v1.2.3


From c11117b634f4f832c4420d3cf41c44227f140ce1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 May 2022 12:48:07 +0200
Subject: ALSA: usb-audio: Refcount multiple accesses on the single clock

When a clock source is connected to multiple nodes / endpoints, the
current USB-audio driver tries to set up at each time one of them is
configured.  Although it reads the current rate and updates only if it
differs, some devices seem unhappy with this behavior and spew the
errors when reading/updating the rate unnecessarily.

This patch tries to reduce the redundant clock setup by introducing a
refcount for each clock source.  When the stream is actually running,
a clock rate is "locked", and it bypasses the clock and/or refuse to
change any longer.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Link: https://lore.kernel.org/r/20220516104807.16482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/card.c     |  1 +
 sound/usb/card.h     |  3 +-
 sound/usb/endpoint.c | 90 ++++++++++++++++++++++++++++++++++++++++++++++------
 sound/usb/usbaudio.h |  1 +
 4 files changed, 85 insertions(+), 10 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/card.c b/sound/usb/card.c
index 376962291c4d..0fff96a5d3ab 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -635,6 +635,7 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 	INIT_LIST_HEAD(&chip->pcm_list);
 	INIT_LIST_HEAD(&chip->ep_list);
 	INIT_LIST_HEAD(&chip->iface_ref_list);
+	INIT_LIST_HEAD(&chip->clock_ref_list);
 	INIT_LIST_HEAD(&chip->midi_list);
 	INIT_LIST_HEAD(&chip->mixer_list);
 
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 87f042d06ce0..ca75f2206170 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -44,6 +44,7 @@ struct audioformat {
 
 struct snd_usb_substream;
 struct snd_usb_iface_ref;
+struct snd_usb_clock_ref;
 struct snd_usb_endpoint;
 struct snd_usb_power_domain;
 
@@ -62,6 +63,7 @@ struct snd_urb_ctx {
 struct snd_usb_endpoint {
 	struct snd_usb_audio *chip;
 	struct snd_usb_iface_ref *iface_ref;
+	struct snd_usb_clock_ref *clock_ref;
 
 	int opened;		/* open refcount; protect with chip->mutex */
 	atomic_t running;	/* running status */
@@ -138,7 +140,6 @@ struct snd_usb_endpoint {
 	unsigned int cur_period_frames;
 	unsigned int cur_period_bytes;
 	unsigned int cur_buffer_periods;
-	unsigned char cur_clock;
 
 	spinlock_t lock;
 	struct list_head list;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 743b8287cfcd..df5a70013a85 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -35,6 +35,14 @@ struct snd_usb_iface_ref {
 	struct list_head list;
 };
 
+/* clock refcounting */
+struct snd_usb_clock_ref {
+	unsigned char clock;
+	atomic_t locked;
+	int rate;
+	struct list_head list;
+};
+
 /*
  * snd_usb_endpoint is a model that abstracts everything related to an
  * USB endpoint and its streaming.
@@ -591,6 +599,25 @@ iface_ref_find(struct snd_usb_audio *chip, int iface)
 	return ip;
 }
 
+/* Similarly, a refcount object for clock */
+static struct snd_usb_clock_ref *
+clock_ref_find(struct snd_usb_audio *chip, int clock)
+{
+	struct snd_usb_clock_ref *ref;
+
+	list_for_each_entry(ref, &chip->clock_ref_list, list)
+		if (ref->clock == clock)
+			return ref;
+
+	ref = kzalloc(sizeof(*ref), GFP_KERNEL);
+	if (!ref)
+		return NULL;
+	ref->clock = clock;
+	atomic_set(&ref->locked, 0);
+	list_add_tail(&ref->list, &chip->clock_ref_list);
+	return ref;
+}
+
 /*
  * Get the existing endpoint object corresponding EP
  * Returns NULL if not present.
@@ -768,6 +795,14 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip,
 			goto unlock;
 		}
 
+		if (fp->protocol != UAC_VERSION_1) {
+			ep->clock_ref = clock_ref_find(chip, fp->clock);
+			if (!ep->clock_ref) {
+				ep = NULL;
+				goto unlock;
+			}
+		}
+
 		ep->cur_audiofmt = fp;
 		ep->cur_channels = fp->channels;
 		ep->cur_rate = params_rate(params);
@@ -777,7 +812,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip,
 		ep->cur_period_frames = params_period_size(params);
 		ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes;
 		ep->cur_buffer_periods = params_periods(params);
-		ep->cur_clock = fp->clock;
 
 		if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC)
 			endpoint_set_syncinterval(chip, ep);
@@ -894,8 +928,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip,
 		ep->altsetting = 0;
 		ep->cur_audiofmt = NULL;
 		ep->cur_rate = 0;
-		ep->cur_clock = 0;
 		ep->iface_ref = NULL;
+		ep->clock_ref = NULL;
 		usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num);
 	}
 	mutex_unlock(&chip->mutex);
@@ -907,6 +941,8 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep)
 	ep->need_setup = true;
 	if (ep->iface_ref)
 		ep->iface_ref->need_setup = true;
+	if (ep->clock_ref)
+		ep->clock_ref->rate = 0;
 }
 
 /*
@@ -1314,6 +1350,33 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip,
 	return 0;
 }
 
+static int init_sample_rate(struct snd_usb_audio *chip,
+			    struct snd_usb_endpoint *ep)
+{
+	struct snd_usb_clock_ref *clock = ep->clock_ref;
+	int err;
+
+	if (clock) {
+		if (atomic_read(&clock->locked))
+			return 0;
+		if (clock->rate == ep->cur_rate)
+			return 0;
+		if (clock->rate && clock->rate != ep->cur_rate) {
+			usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n",
+				      clock->rate, ep->cur_rate, ep->ep_num);
+			return -EINVAL;
+		}
+	}
+
+	err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate);
+	if (err < 0)
+		return err;
+
+	if (clock)
+		clock->rate = ep->cur_rate;
+	return 0;
+}
+
 /*
  * snd_usb_endpoint_configure: Configure the endpoint
  *
@@ -1343,8 +1406,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip,
 		 * to update at each EP configuration
 		 */
 		if (ep->cur_audiofmt->protocol == UAC_VERSION_1) {
-			err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt,
-						       ep->cur_rate);
+			err = init_sample_rate(chip, ep);
 			if (err < 0)
 				goto unlock;
 		}
@@ -1374,7 +1436,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip,
 	if (err < 0)
 		goto unlock;
 
-	err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate);
+	err = init_sample_rate(chip, ep);
 	if (err < 0)
 		goto unlock;
 
@@ -1407,15 +1469,15 @@ unlock:
 /* get the current rate set to the given clock by any endpoint */
 int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock)
 {
-	struct snd_usb_endpoint *ep;
+	struct snd_usb_clock_ref *ref;
 	int rate = 0;
 
 	if (!clock)
 		return 0;
 	mutex_lock(&chip->mutex);
-	list_for_each_entry(ep, &chip->ep_list, list) {
-		if (ep->cur_clock == clock && ep->cur_rate) {
-			rate = ep->cur_rate;
+	list_for_each_entry(ref, &chip->clock_ref_list, list) {
+		if (ref->clock == clock) {
+			rate = ref->rate;
 			break;
 		}
 	}
@@ -1456,6 +1518,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 	if (atomic_inc_return(&ep->running) != 1)
 		return 0;
 
+	if (ep->clock_ref)
+		atomic_inc(&ep->clock_ref->locked);
+
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
 	ep->phase = 0;
@@ -1565,6 +1630,9 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending)
 		if (ep->sync_source)
 			WRITE_ONCE(ep->sync_source->sync_sink, NULL);
 		stop_urbs(ep, false, keep_pending);
+		if (ep->clock_ref)
+			if (!atomic_dec_return(&ep->clock_ref->locked))
+				ep->clock_ref->rate = 0;
 	}
 }
 
@@ -1591,12 +1659,16 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip)
 {
 	struct snd_usb_endpoint *ep, *en;
 	struct snd_usb_iface_ref *ip, *in;
+	struct snd_usb_clock_ref *cp, *cn;
 
 	list_for_each_entry_safe(ep, en, &chip->ep_list, list)
 		kfree(ep);
 
 	list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list)
 		kfree(ip);
+
+	list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list)
+		kfree(ip);
 }
 
 /*
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 044cd7ab27cb..ffbb4b0d09a0 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -45,6 +45,7 @@ struct snd_usb_audio {
 	struct list_head pcm_list;	/* list of pcm streams */
 	struct list_head ep_list;	/* list of audio-related endpoints */
 	struct list_head iface_ref_list; /* list of interface refcounts */
+	struct list_head clock_ref_list; /* list of clock refcounts */
 	int pcm_devs;
 
 	struct list_head midi_list;	/* list of midi interfaces */
-- 
cgit v1.2.3


From 03a8b0df757f1beb21ba1626e23ca7412e48b525 Mon Sep 17 00:00:00 2001
From: Wan Jiabing <wanjiabing@vivo.com>
Date: Wed, 18 May 2022 10:16:16 +0800
Subject: ALSA: usb-audio: Fix wrong kfree issue in snd_usb_endpoint_free_all

Fix following coccicheck error:
./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671.

Here should be 'cp' rather than 'ip'.

Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Signed-off-by: Wan Jiabing <wanjiabing@vivo.com>
Link: https://lore.kernel.org/r/20220518021617.10114-1-wanjiabing@vivo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/endpoint.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/usb')

diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index df5a70013a85..f9c921683948 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -1668,7 +1668,7 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip)
 		kfree(ip);
 
 	list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list)
-		kfree(ip);
+		kfree(cp);
 }
 
 /*
-- 
cgit v1.2.3


From 5ce0b06ae5e69e23142e73c5c3c0260e9f2ccb4b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 21 May 2022 08:46:27 +0200
Subject: ALSA: usb-audio: Workaround for clock setup on TEAC devices

Maris reported that TEAC UD-501 (0644:8043) doesn't work with the
typical "clock source 41 is not valid, cannot use" errors on the
recent kernels.  The currently known workaround so far is to restore
(partially) what we've done unconditionally at the clock setup;
namely, re-setup the USB interface immediately after the clock is
changed.  This patch re-introduces the behavior conditionally for TEAC
devices.

Further notes:
- The USB interface shall be set later in
  snd_usb_endpoint_configure(), but this seems to be too late.
- Even calling  usb_set_interface() right after
  sne_usb_init_sample_rate() doesn't help; so this must be related
  with the clock validation, too.
- The device may still spew the "clock source 41 is not valid" error
  at the first clock setup.  This seems happening at the very first
  try of clock setup, but it disappears at later attempts.
  The error is likely harmless because the driver retries the clock
  setup (such an error is more or less expected on some devices).

Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Reported-and-tested-by: Maris Abele <maris7abele@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220521064627.29292-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/clock.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 4dfe76416794..3c435d379306 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -572,6 +572,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip,
 		/* continue processing */
 	}
 
+	/* FIXME - TEAC devices require the immediate interface setup */
+	if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) {
+		usb_set_interface(chip->dev, fmt->iface, fmt->altsetting);
+		if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY)
+			msleep(50);
+	}
+
 validation:
 	/* validate clock after rate change */
 	if (!uac_clock_source_is_valid(chip, fmt, clock))
-- 
cgit v1.2.3


From 7b0efea4baf02f5e2f89e5f9b75ef891571b45f1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 21 May 2022 08:53:25 +0200
Subject: ALSA: usb-audio: Add missing ep_idx in fixed EP quirks
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The quirk entry for Focusrite Saffire 6 had no proper ep_idx for the
capture endpoint, and this confused the driver, resulting in the
broken sound.  This patch adds the missing ep_idx in the entry.

While we are at it, a couple of other entries (for Digidesign MBox and
MOTU MicroBook II) seem to have the same problem, and those are
covered as well.

Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: André Kapelrud <a.kapelrud@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220521065325.426-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 40a5e3eb4ef2..78eb41b621d6 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2672,6 +2672,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC,
 					.datainterval = 1,
 					.maxpacksize = 0x0126,
@@ -2875,6 +2876,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0x4,
 					.endpoint = 0x81,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC |
 						USB_ENDPOINT_SYNC_ASYNC,
 					.maxpacksize = 0x130,
@@ -3391,6 +3393,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 					.altset_idx = 1,
 					.attributes = 0,
 					.endpoint = 0x03,
+					.ep_idx = 1,
 					.rates = SNDRV_PCM_RATE_96000,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC |
 						   USB_ENDPOINT_SYNC_ASYNC,
-- 
cgit v1.2.3


From 0e85a22d01dfe9ad9a9d9e87cd4a88acce1aad65 Mon Sep 17 00:00:00 2001
From: Craig McLure <craig@mclure.net>
Date: Tue, 24 May 2022 08:21:15 +0200
Subject: ALSA: usb-audio: Configure sync endpoints before data

Devices such as the TC-Helicon GoXLR require the sync endpoint to be
configured in advance of the data endpoint in order for sound output
to work.

This patch simply changes the ordering of EP configuration to resolve
this.

Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215079
Signed-off-by: Craig McLure <craig@mclure.net>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220524062115.25968-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 17 +++++++++++------
 1 file changed, 11 insertions(+), 6 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 6d699065e81a..b470404a5376 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -439,16 +439,21 @@ static int configure_endpoints(struct snd_usb_audio *chip,
 		/* stop any running stream beforehand */
 		if (stop_endpoints(subs, false))
 			sync_pending_stops(subs);
+		if (subs->sync_endpoint) {
+			err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
+			if (err < 0)
+				return err;
+		}
 		err = snd_usb_endpoint_configure(chip, subs->data_endpoint);
 		if (err < 0)
 			return err;
 		snd_usb_set_format_quirk(subs, subs->cur_audiofmt);
-	}
-
-	if (subs->sync_endpoint) {
-		err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
-		if (err < 0)
-			return err;
+	} else {
+		if (subs->sync_endpoint) {
+			err = snd_usb_endpoint_configure(chip, subs->sync_endpoint);
+			if (err < 0)
+				return err;
+		}
 	}
 
 	return 0;
-- 
cgit v1.2.3


From 4c691a287d4ee0c308708c1d6f9e0cc7513463f8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 25 May 2022 14:20:18 +0200
Subject: ALSA: usb-audio: Add mixer mapping for Gigabyte B450/550 Mobos

This patch implements a static mapping for Gigabyte B450/550 Mobos so
that the mixer elements appear reasonably and jack detections work
properly.

Reported-and-tested-by: Brock Szuszczewicz <brock@system76.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215988
Link: https://lore.kernel.org/r/20220525122018.3299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_maps.c | 30 ++++++++++++++++++++++++++++++
 1 file changed, 30 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 7ef7a8abcc2b..3c795675f048 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -439,6 +439,31 @@ static const struct usbmix_name_map msi_mpg_x570s_carbon_max_wifi_alc4080_map[]
 	{}
 };
 
+/* Gigabyte B450/550 Mobo */
+static const struct usbmix_name_map gigabyte_b450_map[] = {
+	{ 24, NULL },			/* OT, IEC958?, disabled */
+	{ 21, "Speaker" },		/* OT */
+	{ 29, "Speaker Playback" },	/* FU */
+	{ 22, "Headphone" },		/* OT */
+	{ 30, "Headphone Playback" },	/* FU */
+	{ 11, "Line" },			/* IT */
+	{ 27, "Line Capture" },		/* FU */
+	{ 12, "Mic" },			/* IT */
+	{ 28, "Mic Capture" },		/* FU */
+	{ 9, "Front Mic" },		/* IT */
+	{ 25, "Front Mic Capture" },	/* FU */
+	{}
+};
+
+static const struct usbmix_connector_map gigabyte_b450_connector_map[] = {
+	{ 13, 21 },	/* Speaker */
+	{ 14, 22 },	/* Headphone */
+	{ 19, 11 },	/* Line */
+	{ 20, 12 },	/* Mic */
+	{ 17, 9 },	/* Front Mic */
+	{}
+};
+
 /*
  * Control map entries
  */
@@ -581,6 +606,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = trx40_mobo_map,
 		.connector_map = trx40_mobo_connector_map,
 	},
+	{	/* Gigabyte B450/550 Mobo */
+		.id = USB_ID(0x0414, 0xa00d),
+		.map = gigabyte_b450_map,
+		.connector_map = gigabyte_b450_connector_map,
+	},
 	{	/* ASUS ROG Zenith II */
 		.id = USB_ID(0x0b05, 0x1916),
 		.map = asus_rog_map,
-- 
cgit v1.2.3


From 0125de38122f0f66bf61336158d12a1aabfe6425 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 25 May 2022 15:12:03 +0200
Subject: ALSA: usb-audio: Cancel pending work at closing a MIDI substream

At closing a USB MIDI output substream, there might be still a pending
work, which would eventually access the rawmidi runtime object that is
being released.  For fixing the race, make sure to cancel the pending
work at closing.

Reported-by: syzbot+6912c9592caca7ca0e7d@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000e7e75005dfd07cf6@google.com
Link: https://lore.kernel.org/r/20220525131203.11299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/midi.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/usb')

diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7c6ca2b433a5..344fbeadf161 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1145,6 +1145,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
 
 static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
 {
+	struct usbmidi_out_port *port = substream->runtime->private_data;
+
+	cancel_work_sync(&port->ep->work);
 	return substream_open(substream, 0, 0);
 }
 
-- 
cgit v1.2.3


From 3753fcc22974affa26160ce1c46a6ebaaaa86758 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 31 May 2022 15:07:49 +0200
Subject: ALSA: usb-audio: Optimize TEAC clock quirk

Maris found out that the quirk for TEAC devices to work around the
clock setup is needed to apply only when the base clock is changed,
e.g. from 48000-based clocks (48000, 96000, 192000, 384000) to
44100-based clocks (44100, 88200, 176400, 352800), or vice versa,
while switching to another clock with the same base clock doesn't need
the (forcible) interface setup.

This patch implements the optimization for the TEAC clock quirk to
avoid the unnecessary interface re-setup.

Fixes: 5ce0b06ae5e6 ("ALSA: usb-audio: Workaround for clock setup on TEAC devices")
Reported-by: Maris Abele <maris7abele@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220531130749.30357-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/clock.c | 12 ++++++++----
 1 file changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound/usb')

diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 3c435d379306..33db334e6556 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -573,10 +573,14 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip,
 	}
 
 	/* FIXME - TEAC devices require the immediate interface setup */
-	if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) {
-		usb_set_interface(chip->dev, fmt->iface, fmt->altsetting);
-		if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY)
-			msleep(50);
+	if (USB_ID_VENDOR(chip->usb_id) == 0x0644) {
+		bool cur_base_48k = (rate % 48000 == 0);
+		bool prev_base_48k = (prev_rate % 48000 == 0);
+		if (cur_base_48k != prev_base_48k) {
+			usb_set_interface(chip->dev, fmt->iface, fmt->altsetting);
+			if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY)
+				msleep(50);
+		}
 	}
 
 validation:
-- 
cgit v1.2.3