<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-stable.git/include/sound, branch v6.1.136</title>
<subtitle>Linux kernel stable tree</subtitle>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/'/>
<entry>
<title>ASoC: ops: Consistently treat platform_max as control value</title>
<updated>2025-03-28T20:58:57+00:00</updated>
<author>
<name>Charles Keepax</name>
<email>ckeepax@opensource.cirrus.com</email>
</author>
<published>2025-02-28T15:14:56+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=694110bc2407a61f02a770cbb5f39b51e4ec77c6'/>
<id>694110bc2407a61f02a770cbb5f39b51e4ec77c6</id>
<content type='text'>
[ Upstream commit 0eba2a7e858907a746ba69cd002eb9eb4dbd7bf3 ]

This reverts commit 9bdd10d57a88 ("ASoC: ops: Shift tested values in
snd_soc_put_volsw() by +min"), and makes some additional related
updates.

There are two ways the platform_max could be interpreted; the maximum
register value, or the maximum value the control can be set to. The
patch moved from treating the value as a control value to a register
one. When the patch was applied it was technically correct as
snd_soc_limit_volume() also used the register interpretation. However,
even then most of the other usages treated platform_max as a
control value, and snd_soc_limit_volume() has since been updated to
also do so in commit fb9ad24485087 ("ASoC: ops: add correct range
check for limiting volume"). That patch however, missed updating
snd_soc_put_volsw() back to the control interpretation, and fixing
snd_soc_info_volsw_range(). The control interpretation makes more
sense as limiting is typically done from the machine driver, so it is
appropriate to use the customer facing representation rather than the
internal codec representation. Update all the code to consistently use
this interpretation of platform_max.

Finally, also add some comments to the soc_mixer_control struct to
hopefully avoid further patches switching between the two approaches.

Fixes: fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume")
Signed-off-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit 0eba2a7e858907a746ba69cd002eb9eb4dbd7bf3 ]

This reverts commit 9bdd10d57a88 ("ASoC: ops: Shift tested values in
snd_soc_put_volsw() by +min"), and makes some additional related
updates.

There are two ways the platform_max could be interpreted; the maximum
register value, or the maximum value the control can be set to. The
patch moved from treating the value as a control value to a register
one. When the patch was applied it was technically correct as
snd_soc_limit_volume() also used the register interpretation. However,
even then most of the other usages treated platform_max as a
control value, and snd_soc_limit_volume() has since been updated to
also do so in commit fb9ad24485087 ("ASoC: ops: add correct range
check for limiting volume"). That patch however, missed updating
snd_soc_put_volsw() back to the control interpretation, and fixing
snd_soc_info_volsw_range(). The control interpretation makes more
sense as limiting is typically done from the machine driver, so it is
appropriate to use the customer facing representation rather than the
internal codec representation. Update all the code to consistently use
this interpretation of platform_max.

Finally, also add some comments to the soc_mixer_control struct to
hopefully avoid further patches switching between the two approaches.

Fixes: fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume")
Signed-off-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: dmaengine: Synchronize dma channel after drop()</title>
<updated>2024-07-25T07:49:14+00:00</updated>
<author>
<name>Jai Luthra</name>
<email>j-luthra@ti.com</email>
</author>
<published>2024-06-11T12:32:55+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=3e25eb518ffef5d1e5d8a044d1e4f2821769aacb'/>
<id>3e25eb518ffef5d1e5d8a044d1e4f2821769aacb</id>
<content type='text'>
[ Upstream commit e8343410ddf08fc36a9b9cc7c51a4e53a262d4c6 ]

Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.

In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.

Reviewed-by: Peter Ujfalusi &lt;peter.ujfalusi@gmail.com&gt;
Signed-off-by: Jai Luthra &lt;j-luthra@ti.com&gt;
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit e8343410ddf08fc36a9b9cc7c51a4e53a262d4c6 ]

Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.

In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.

Reviewed-by: Peter Ujfalusi &lt;peter.ujfalusi@gmail.com&gt;
Signed-off-by: Jai Luthra &lt;j-luthra@ti.com&gt;
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: SOF: Pass PCI SSID to machine driver</title>
<updated>2023-11-28T17:06:58+00:00</updated>
<author>
<name>Richard Fitzgerald</name>
<email>rf@opensource.cirrus.com</email>
</author>
<published>2023-09-12T16:32:05+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=62c65e799fb47db046ffcf12e0b46d390260c599'/>
<id>62c65e799fb47db046ffcf12e0b46d390260c599</id>
<content type='text'>
[ Upstream commit ba2de401d32625fe538d3f2c00ca73740dd2d516 ]

Pass the PCI SSID of the audio interface through to the machine driver.
This allows the machine driver to use the SSID to uniquely identify the
specific hardware configuration and apply any platform-specific
configuration.

struct snd_sof_pdata is passed around inside the SOF code, but it then
passes configuration information to the machine driver through
struct snd_soc_acpi_mach and struct snd_soc_acpi_mach_params. So SSID
information has been added to both snd_sof_pdata and
snd_soc_acpi_mach_params.

PCI does not define 0x0000 as an invalid value so we can't use zero to
indicate that the struct member was not written. Instead a flag is
included to indicate that a value has been written to the
subsystem_vendor and subsystem_device members.

sof_pci_probe() creates the struct snd_sof_pdata. It is passed a struct
pci_dev so it can fill in the SSID value.

sof_machine_check() finds the appropriate struct snd_soc_acpi_mach. It
copies the SSID information across to the struct snd_soc_acpi_mach_params.
This done before calling any custom set_mach_params() so that it could be
used by the set_mach_params() callback to apply variant params.

The machine driver receives the struct snd_soc_acpi_mach as its
platform_data.

Signed-off-by: Richard Fitzgerald &lt;rf@opensource.cirrus.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230912163207.3498161-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit ba2de401d32625fe538d3f2c00ca73740dd2d516 ]

Pass the PCI SSID of the audio interface through to the machine driver.
This allows the machine driver to use the SSID to uniquely identify the
specific hardware configuration and apply any platform-specific
configuration.

struct snd_sof_pdata is passed around inside the SOF code, but it then
passes configuration information to the machine driver through
struct snd_soc_acpi_mach and struct snd_soc_acpi_mach_params. So SSID
information has been added to both snd_sof_pdata and
snd_soc_acpi_mach_params.

PCI does not define 0x0000 as an invalid value so we can't use zero to
indicate that the struct member was not written. Instead a flag is
included to indicate that a value has been written to the
subsystem_vendor and subsystem_device members.

sof_pci_probe() creates the struct snd_sof_pdata. It is passed a struct
pci_dev so it can fill in the SSID value.

sof_machine_check() finds the appropriate struct snd_soc_acpi_mach. It
copies the SSID information across to the struct snd_soc_acpi_mach_params.
This done before calling any custom set_mach_params() so that it could be
used by the set_mach_params() callback to apply variant params.

The machine driver receives the struct snd_soc_acpi_mach as its
platform_data.

Signed-off-by: Richard Fitzgerald &lt;rf@opensource.cirrus.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230912163207.3498161-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: soc-card: Add storage for PCI SSID</title>
<updated>2023-11-28T17:06:58+00:00</updated>
<author>
<name>Richard Fitzgerald</name>
<email>rf@opensource.cirrus.com</email>
</author>
<published>2023-09-12T16:32:04+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=14107cbeb5f709d0bb25e0d334a225cfb5280dce'/>
<id>14107cbeb5f709d0bb25e0d334a225cfb5280dce</id>
<content type='text'>
[ Upstream commit 47f56e38a199bd45514b8e0142399cba4feeaf1a ]

Add members to struct snd_soc_card to store the PCI subsystem ID (SSID)
of the soundcard.

The PCI specification provides two registers to store a vendor-specific
SSID that can be read by drivers to uniquely identify a particular
"soundcard". This is defined in the PCI specification to distinguish
products that use the same silicon (and therefore have the same silicon
ID) so that product-specific differences can be applied.

PCI only defines 0xFFFF as an invalid value. 0x0000 is not defined as
invalid. So the usual pattern of zero-filling the struct and then
assuming a zero value unset will not work. A flag is included to
indicate when the SSID information has been filled in.

Unlike DMI information, which has a free-format entirely up to the vendor,
the PCI SSID has a strictly defined format and a registry of vendor IDs.

It is usual in Windows drivers that the SSID is used as the sole identifier
of the specific end-product and the Windows driver contains tables mapping
that to information about the hardware setup, rather than using ACPI
properties.

This SSID is important information for ASoC components that need to apply
hardware-specific configuration on PCI-based systems.

As the SSID is a generic part of the PCI specification and is treated as
identifying the "soundcard", it is reasonable to include this information
in struct snd_soc_card, instead of components inventing their own custom
ways to pass this information around.

Signed-off-by: Richard Fitzgerald &lt;rf@opensource.cirrus.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230912163207.3498161-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit 47f56e38a199bd45514b8e0142399cba4feeaf1a ]

Add members to struct snd_soc_card to store the PCI subsystem ID (SSID)
of the soundcard.

The PCI specification provides two registers to store a vendor-specific
SSID that can be read by drivers to uniquely identify a particular
"soundcard". This is defined in the PCI specification to distinguish
products that use the same silicon (and therefore have the same silicon
ID) so that product-specific differences can be applied.

PCI only defines 0xFFFF as an invalid value. 0x0000 is not defined as
invalid. So the usual pattern of zero-filling the struct and then
assuming a zero value unset will not work. A flag is included to
indicate when the SSID information has been filled in.

Unlike DMI information, which has a free-format entirely up to the vendor,
the PCI SSID has a strictly defined format and a registry of vendor IDs.

It is usual in Windows drivers that the SSID is used as the sole identifier
of the specific end-product and the Windows driver contains tables mapping
that to information about the hardware setup, rather than using ACPI
properties.

This SSID is important information for ASoC components that need to apply
hardware-specific configuration on PCI-based systems.

As the SSID is a generic part of the PCI specification and is treated as
identifying the "soundcard", it is reasonable to include this information
in struct snd_soc_card, instead of components inventing their own custom
ways to pass this information around.

Signed-off-by: Richard Fitzgerald &lt;rf@opensource.cirrus.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230912163207.3498161-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: Intel: avs: Account for UID of ACPI device</title>
<updated>2023-06-21T14:00:53+00:00</updated>
<author>
<name>Cezary Rojewski</name>
<email>cezary.rojewski@intel.com</email>
</author>
<published>2023-05-19T20:17:09+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=5daa27bcb31d731258b39497dbd0a7949eee75b9'/>
<id>5daa27bcb31d731258b39497dbd0a7949eee75b9</id>
<content type='text'>
[ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ]

Configurations with multiple codecs attached to the platform are
supported but only if each from the set is different. Add new field
representing the 'Unique ID' so that codecs that share Vendor and Part
IDs can be differentiated and thus enabling support for such
configurations.

Signed-off-by: Cezary Rojewski &lt;cezary.rojewski@intel.com&gt;
Signed-off-by: Amadeusz Sławiński &lt;amadeuszx.slawinski@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ]

Configurations with multiple codecs attached to the platform are
supported but only if each from the set is different. Add new field
representing the 'Unique ID' so that codecs that share Vendor and Part
IDs can be differentiated and thus enabling support for such
configurations.

Signed-off-by: Cezary Rojewski &lt;cezary.rojewski@intel.com&gt;
Signed-off-by: Amadeusz Sławiński &lt;amadeuszx.slawinski@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: soc-pcm: test if a BE can be prepared</title>
<updated>2023-06-21T14:00:53+00:00</updated>
<author>
<name>Ranjani Sridharan</name>
<email>ranjani.sridharan@linux.intel.com</email>
</author>
<published>2023-05-17T18:57:31+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=c33fded7f17f6102a4d7d77f826d0eba5da9b986'/>
<id>c33fded7f17f6102a4d7d77f826d0eba5da9b986</id>
<content type='text'>
[ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ]

In the BE hw_params configuration, the existing code checks if any of the
existing FEs are prepared, running, paused or suspended - and skips the
configuration in those cases. This allows multiple calls of hw_params
which the ALSA state machine supports.

This check is not handled for the prepare stage, which can lead to the
same BE being prepared multiple times. This patch adds a check similar to
that of the hw_params, with the main difference being that the suspended
state is allowed: the ALSA state machine allows a transition from
suspended to prepared with hw_params skipped.

This problem was detected on Intel IPC4/SoundWire devices, where the BE
dailink .prepare stage is used to configure the SoundWire stream with a
bank switch. Multiple .prepare calls lead to conflicts with the .trigger
operation with IPC4 configurations. This problem was not detected earlier
on Intel devices, HDaudio BE dailinks detect that the link is already
prepared and skip the configuration, and for IPC3 devices there is no BE
trigger.

Link: https://github.com/thesofproject/sof/issues/7596
Signed-off-by: Ranjani Sridharan &lt;ranjani.sridharan@linux.intel.com
Signed-off-by: Bard Liao &lt;yung-chuan.liao@linux.intel.com
Signed-off-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ]

In the BE hw_params configuration, the existing code checks if any of the
existing FEs are prepared, running, paused or suspended - and skips the
configuration in those cases. This allows multiple calls of hw_params
which the ALSA state machine supports.

This check is not handled for the prepare stage, which can lead to the
same BE being prepared multiple times. This patch adds a check similar to
that of the hw_params, with the main difference being that the suspended
state is allowed: the ALSA state machine allows a transition from
suspended to prepared with hw_params skipped.

This problem was detected on Intel IPC4/SoundWire devices, where the BE
dailink .prepare stage is used to configure the SoundWire stream with a
bank switch. Multiple .prepare calls lead to conflicts with the .trigger
operation with IPC4 configurations. This problem was not detected earlier
on Intel devices, HDaudio BE dailinks detect that the link is already
prepared and skip the configuration, and for IPC3 devices there is no BE
trigger.

Link: https://github.com/thesofproject/sof/issues/7596
Signed-off-by: Ranjani Sridharan &lt;ranjani.sridharan@linux.intel.com
Signed-off-by: Bard Liao &lt;yung-chuan.liao@linux.intel.com
Signed-off-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: amd: fix ACP version typo mistake</title>
<updated>2023-05-11T14:02:59+00:00</updated>
<author>
<name>syed saba kareem</name>
<email>syed.sabakareem@amd.com</email>
</author>
<published>2022-11-04T12:09:07+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=f9dc736e686cfb2bd961c5abc09b23e565ace05d'/>
<id>f9dc736e686cfb2bd961c5abc09b23e565ace05d</id>
<content type='text'>
commit 4b19211435950a78af032c26ad64a5268e6012be upstream.

Pink Sardine is based on ACP6.3 architecture.
This patch fixes the typo mistake acp6.2 -&gt; acp6.3

Signed-off-by: syed saba kareem &lt;syed.sabakareem@amd.com&gt;
Link: https://lore.kernel.org/r/20221104121001.207992-1-Syed.SabaKareem@amd.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Cc: Mario Limonciello &lt;mario.limonciello@amd.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
commit 4b19211435950a78af032c26ad64a5268e6012be upstream.

Pink Sardine is based on ACP6.3 architecture.
This patch fixes the typo mistake acp6.2 -&gt; acp6.3

Signed-off-by: syed saba kareem &lt;syed.sabakareem@amd.com&gt;
Link: https://lore.kernel.org/r/20221104121001.207992-1-Syed.SabaKareem@amd.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Cc: Mario Limonciello &lt;mario.limonciello@amd.com&gt;
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared</title>
<updated>2023-03-10T08:33:23+00:00</updated>
<author>
<name>Lucas Tanure</name>
<email>lucas.tanure@collabora.com</email>
</author>
<published>2023-02-15T13:28:51+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=1c3609ee824edc8dafb5e140be47cb7358ef4f0e'/>
<id>1c3609ee824edc8dafb5e140be47cb7358ef4f0e</id>
<content type='text'>
[ Upstream commit fdff966bfde7cf0c85562d2bfb1ff1ba83da5f7b ]

Add struct snd_pcm_substream forward declaration

Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI")
Signed-off-by: Lucas Tanure &lt;lucas.tanure@collabora.com&gt;
Reviewed-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Reviewed-by: AngeloGioacchino Del Regno &lt;angelogioacchino.delregno@collabora.com&gt;
Link: https://lore.kernel.org/r/20230215132851.1626881-1-lucas.tanure@collabora.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit fdff966bfde7cf0c85562d2bfb1ff1ba83da5f7b ]

Add struct snd_pcm_substream forward declaration

Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI")
Signed-off-by: Lucas Tanure &lt;lucas.tanure@collabora.com&gt;
Reviewed-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Reviewed-by: AngeloGioacchino Del Regno &lt;angelogioacchino.delregno@collabora.com&gt;
Link: https://lore.kernel.org/r/20230215132851.1626881-1-lucas.tanure@collabora.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: hda: Fix the control element identification for multiple codecs</title>
<updated>2023-03-10T08:33:20+00:00</updated>
<author>
<name>Jaroslav Kysela</name>
<email>perex@perex.cz</email>
</author>
<published>2023-02-02T09:20:13+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=f716ca292a7d1808d4135273bbdca093da195831'/>
<id>f716ca292a7d1808d4135273bbdca093da195831</id>
<content type='text'>
[ Upstream commit d045bceff5a904bd79d71dede9f927c00ce4906f ]

Some motherboards have multiple HDA codecs connected to the serial bus.
The current code may create multiple mixer controls with the almost
identical identification.

The current code use id.device field from the control element structure
to store the codec address to avoid such clashes for multiple codecs.
Unfortunately, the user space do not handle this correctly. For mixer
controls, only name and index are used for the identifiers.

This patch fixes this problem to compose the index using the codec
address as an offset in case, when the control already exists. It is
really unlikely that one codec will create 10 similar controls.

This patch adds new kernel module parameter 'ctl_dev_id' to allow
select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig
option sets the default value.

BugLink: https://github.com/alsa-project/alsa-lib/issues/294
BugLink: https://github.com/alsa-project/alsa-lib/issues/205
Fixes: 54d174031576 ("[ALSA] hda-codec - Fix connection list parsing")
Fixes: 1afe206ab699 ("ALSA: hda - Try to find an empty control index when it's occupied")
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
Link: https://lore.kernel.org/r/20230202092013.4066998-1-perex@perex.cz
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit d045bceff5a904bd79d71dede9f927c00ce4906f ]

Some motherboards have multiple HDA codecs connected to the serial bus.
The current code may create multiple mixer controls with the almost
identical identification.

The current code use id.device field from the control element structure
to store the codec address to avoid such clashes for multiple codecs.
Unfortunately, the user space do not handle this correctly. For mixer
controls, only name and index are used for the identifiers.

This patch fixes this problem to compose the index using the codec
address as an offset in case, when the control already exists. It is
really unlikely that one codec will create 10 similar controls.

This patch adds new kernel module parameter 'ctl_dev_id' to allow
select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig
option sets the default value.

BugLink: https://github.com/alsa-project/alsa-lib/issues/294
BugLink: https://github.com/alsa-project/alsa-lib/issues/205
Fixes: 54d174031576 ("[ALSA] hda-codec - Fix connection list parsing")
Fixes: 1afe206ab699 ("ALSA: hda - Try to find an empty control index when it's occupied")
Signed-off-by: Jaroslav Kysela &lt;perex@perex.cz&gt;
Link: https://lore.kernel.org/r/20230202092013.4066998-1-perex@perex.cz
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>ALSA: hda/hdmi: fix stream-id config keep-alive for rt suspend</title>
<updated>2022-12-31T12:33:07+00:00</updated>
<author>
<name>Kai Vehmanen</name>
<email>kai.vehmanen@linux.intel.com</email>
</author>
<published>2022-12-09T10:18:22+00:00</published>
<link rel='alternate' type='text/html' href='https://git.tavy.me/linux-stable.git/commit/?id=f64bb48f27fbe789e9d66930008267cfccccac1e'/>
<id>f64bb48f27fbe789e9d66930008267cfccccac1e</id>
<content type='text'>
[ Upstream commit ee0b089d660021792e4ab4dda191b097ce1e964f ]

When the new style KAE keep-alive implementation is used on compatible
Intel hardware, the clocks are maintained when codec is in D3. The
generic code in hda_cleanup_all_streams() can however interfere with
generation of audio samples in this mode, by setting the stream and
channel ids to zero.

To get full benefit of the keepalive, set the new
no_stream_clean_at_suspend quirk bit on affected Intel hardware. When
this bit is set, stream cleanup is skipped in hda_call_codec_suspend().

Special handling is needed for the case when system goes to suspend. The
stream id programming can be lost in this case. This will also cause
codec-&gt;cvt_setups to be out of sync. Handle this by implementing custom
suspend/resume handlers. If keep-alive is active for any converter, set
the quirk flags no_stream_clean_at_suspend and forced_resume. Upon
resume, keepalive programming is restored if needed.

Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Signed-off-by: Kai Vehmanen &lt;kai.vehmanen@linux.intel.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20221209101822.3893675-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
[ Upstream commit ee0b089d660021792e4ab4dda191b097ce1e964f ]

When the new style KAE keep-alive implementation is used on compatible
Intel hardware, the clocks are maintained when codec is in D3. The
generic code in hda_cleanup_all_streams() can however interfere with
generation of audio samples in this mode, by setting the stream and
channel ids to zero.

To get full benefit of the keepalive, set the new
no_stream_clean_at_suspend quirk bit on affected Intel hardware. When
this bit is set, stream cleanup is skipped in hda_call_codec_suspend().

Special handling is needed for the case when system goes to suspend. The
stream id programming can be lost in this case. This will also cause
codec-&gt;cvt_setups to be out of sync. Handle this by implementing custom
suspend/resume handlers. If keep-alive is active for any converter, set
the quirk flags no_stream_clean_at_suspend and forced_resume. Upon
resume, keepalive programming is restored if needed.

Fixes: 15175a4f2bbb ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Signed-off-by: Kai Vehmanen &lt;kai.vehmanen@linux.intel.com&gt;
Reviewed-by: Pierre-Louis Bossart &lt;pierre-louis.bossart@linux.intel.com&gt;
Link: https://lore.kernel.org/r/20221209101822.3893675-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</pre>
</div>
</content>
</entry>
</feed>
