[PATCH AUTOSEL 6.11 02/32] ASoC: Intel: avs: Update stream status in a separate thread

[PATCH AUTOSEL 6.11 02/32] ASoC: Intel: avs: Update stream status in a separate thread

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 0dbb186c3510cad4e9f443e801bf2e6ab5770c00 ]

Function snd_pcm_period_elapsed() is part of sequence servicing HDAudio
stream IRQs. It's called under Global Interrupt Enable (GIE) disabled -
no HDAudio interrupts will be raised. At the same time, the function may
end up calling __snd_pcm_xrun() or snd_pcm_drain_done(). On the
avs-driver side, this translates to IPCs and as GIE is disabled, these
will never complete successfully.

Improve system stability by scheduling stream-IRQ handling in a separate
thread.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20241008083758.756578-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/core.c |  3 ++-
 sound/soc/intel/avs/pcm.c  | 19 +++++++++++++++++++
 sound/soc/intel/avs/pcm.h  | 16 ++++++++++++++++
 3 files changed, 37 insertions(+), 1 deletion(-)
 create mode 100644 sound/soc/intel/avs/pcm.h

diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c
index f2dc82a2abc71..4d1e6c84918c6 100644
--- a/sound/soc/intel/avs/core.c
+++ b/sound/soc/intel/avs/core.c
@@ -28,6 +28,7 @@
 #include "avs.h"
 #include "cldma.h"
 #include "messages.h"
+#include "pcm.h"
 
 static u32 pgctl_mask = AZX_PGCTL_LSRMD_MASK;
 module_param(pgctl_mask, uint, 0444);
@@ -247,7 +248,7 @@ static void hdac_stream_update_pos(struct hdac_stream *stream, u64 buffer_size)
 static void hdac_update_stream(struct hdac_bus *bus, struct hdac_stream *stream)
 {
 	if (stream->substream) {
-		snd_pcm_period_elapsed(stream->substream);
+		avs_period_elapsed(stream->substream);
 	} else if (stream->cstream) {
 		u64 buffer_size = stream->cstream->runtime->buffer_size;
 
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index c76b86254a8b4..37b1880c81141 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -16,6 +16,7 @@
 #include <sound/soc-component.h>
 #include "avs.h"
 #include "path.h"
+#include "pcm.h"
 #include "topology.h"
 #include "../../codecs/hda.h"
 
@@ -30,6 +31,7 @@ struct avs_dma_data {
 		struct hdac_ext_stream *host_stream;
 	};
 
+	struct work_struct period_elapsed_work;
 	struct snd_pcm_substream *substream;
 };
 
@@ -56,6 +58,22 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction)
 	return dw->priv;
 }
 
+static void avs_period_elapsed_work(struct work_struct *work)
+{
+	struct avs_dma_data *data = container_of(work, struct avs_dma_data, period_elapsed_work);
+
+	snd_pcm_period_elapsed(data->substream);
+}
+
+void avs_period_elapsed(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
+	struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(rtd, 0);
+	struct avs_dma_data *data = snd_soc_dai_get_dma_data(dai, substream);
+
+	schedule_work(&data->period_elapsed_work);
+}
+
 static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
@@ -77,6 +95,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d
 	data->substream = substream;
 	data->template = template;
 	data->adev = adev;
+	INIT_WORK(&data->period_elapsed_work, avs_period_elapsed_work);
 	snd_soc_dai_set_dma_data(dai, substream, data);
 
 	if (rtd->dai_link->ignore_suspend)
diff --git a/sound/soc/intel/avs/pcm.h b/sound/soc/intel/avs/pcm.h
new file mode 100644
index 0000000000000..0f3615c903982
--- /dev/null
+++ b/sound/soc/intel/avs/pcm.h
@@ -0,0 +1,16 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2024 Intel Corporation
+ *
+ * Authors: Cezary Rojewski <cezary.rojewski@intel.com>
+ *          Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
+ */
+
+#ifndef __SOUND_SOC_INTEL_AVS_PCM_H
+#define __SOUND_SOC_INTEL_AVS_PCM_H
+
+#include <sound/pcm.h>
+
+void avs_period_elapsed(struct snd_pcm_substream *substream);
+
+#endif
-- 
2.43.0

[PATCH AUTOSEL 6.11 03/32] ASoC: codecs: Fix error handling in aw_dev_get_dsp_status function

[Sasha Levin]

From: Zhu Jun <zhujun2@cmss.chinamobile.com>

[ Upstream commit 251ce34a446ef0e1d6acd65cf5947abd5d10b8b6 ]

Added proper error handling for register value check that
return -EPERM when register value does not meet expected condition

Signed-off-by: Zhu Jun <zhujun2@cmss.chinamobile.com>
Link: https://patch.msgid.link/20241009073938.7472-1-zhujun2@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/aw88399.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/aw88399.c b/sound/soc/codecs/aw88399.c
index 8dc2b8aa6832d..bba59885242d0 100644
--- a/sound/soc/codecs/aw88399.c
+++ b/sound/soc/codecs/aw88399.c
@@ -656,7 +656,7 @@ static int aw_dev_get_dsp_status(struct aw_device *aw_dev)
 	if (ret)
 		return ret;
 	if (!(reg_val & (~AW88399_WDT_CNT_MASK)))
-		ret = -EPERM;
+		return -EPERM;
 
 	return 0;
 }
-- 
2.43.0

[PATCH AUTOSEL 6.11 04/32] ASoC: amd: yc: Add quirk for ASUS Vivobook S15 M3502RA

[Sasha Levin]

From: Christian Heusel <christian@heusel.eu>

[ Upstream commit 182fff3a2aafe4e7f3717a0be9df2fe2ed1a77de ]

As reported the builtin microphone doesn't work on the ASUS Vivobook
model S15 OLED M3502RA. Therefore add a quirk for it to make it work.

Link: https://bugzilla.kernel.org/show_bug.cgi?id=219345
Signed-off-by: Christian Heusel <christian@heusel.eu>
Link: https://patch.msgid.link/20241010-bugzilla-219345-asus-vivobook-v1-1-3bb24834e2c3@heusel.eu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/amd/yc/acp6x-mach.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 06349bf0b6587..e353542dcf636 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -339,6 +339,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M7600RE"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "M3502RA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
-- 
2.43.0

[PATCH AUTOSEL 6.11 05/32] ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler

[Sasha Levin]

From: Shengjiu Wang <shengjiu.wang@nxp.com>

[ Upstream commit 54c805c1eb264c839fa3027d0073bb7f323b0722 ]

Irq handler need to be executed as fast as possible, so
the log in irq handler is better to use dev_dbg which needs
to be enabled when debugging.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Iuliana Prodan <iuliana.prodan@nxp.com>
Link: https://patch.msgid.link/1728622433-2873-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index d0d8a01da9bdd..0cf9484183d43 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -119,10 +119,10 @@ static irqreturn_t esai_isr(int irq, void *devid)
 		dev_dbg(&pdev->dev, "isr: Transmission Initialized\n");
 
 	if (esr & ESAI_ESR_RFF_MASK)
-		dev_warn(&pdev->dev, "isr: Receiving overrun\n");
+		dev_dbg(&pdev->dev, "isr: Receiving overrun\n");
 
 	if (esr & ESAI_ESR_TFE_MASK)
-		dev_warn(&pdev->dev, "isr: Transmission underrun\n");
+		dev_dbg(&pdev->dev, "isr: Transmission underrun\n");
 
 	if (esr & ESAI_ESR_TLS_MASK)
 		dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n");
-- 
2.43.0

[PATCH AUTOSEL 6.11 06/32] ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs

[Sasha Levin]

From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>

[ Upstream commit 9822b4c90d77e3c6555fb21c459c4a61c6a8619f ]

For aggregated DAIs, the node ID is set to the group_id during the DAI
widget's ipc_prepare op. With the current logic, setting the dai_index
for node_id in the dai_config is redundant as it will be overwritten
with the group_id anyway. Removing it will also prevent any accidental
clearing/resetting of the group_id for aggregated DAIs due to the
dai_config calls could that happen before the allocated group_id is
freed.

Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/ipc4-topology.c | 15 +++++++++++++--
 1 file changed, 13 insertions(+), 2 deletions(-)

diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 87be7f16e8c2b..240fee2166d12 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -3129,9 +3129,20 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		 * group_id during copier's ipc_prepare op.
 		 */
 		if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+			struct sof_ipc4_alh_configuration_blob *blob;
+
+			blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config;
 			ipc4_copier->dai_index = data->dai_node_id;
-			copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
-			copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id);
+
+			/*
+			 * no need to set the node_id for aggregated DAI's. These will be assigned
+			 * a group_id during widget ipc_prepare
+			 */
+			if (blob->alh_cfg.device_count == 1) {
+				copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+				copier_data->gtw_cfg.node_id |=
+					SOF_IPC4_NODE_INDEX(data->dai_node_id);
+			}
 		}
 
 		break;
-- 
2.43.0

[PATCH AUTOSEL 6.11 07/32] ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's

[Sasha Levin]

From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>

[ Upstream commit 6e38a7e098d32d128b00b42a536151de9ea1340b ]

When a PCM is restarted after a snd_pcm_drain/snd_pcm_drop(), the prepare
callback will be invoked and the hw_params will be set again. For the
HDA DAI's, the hw_params function handles this case already but not for
the non-HDA DAI's. So, add the check for link_prepared to verify if the
hw_params should be done again or not. Additionally, for SDW DAI's reset
the PCMSyCM registers as would be done in the case of a start after a
hw_free.

Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/hda-dai.c | 36 +++++++++++++++++++++++++++++++----
 1 file changed, 32 insertions(+), 4 deletions(-)

diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 1c823f9eea570..8cccf38967e72 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -370,6 +370,13 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	sdev = widget_to_sdev(w);
+	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
+
+	/* nothing more to do if the link is already prepared */
+	if (hext_stream && hext_stream->link_prepared)
+		return 0;
+
 	/* use HDaudio stream handling */
 	ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags);
 	if (ret < 0) {
@@ -377,7 +384,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	sdev = widget_to_sdev(w);
 	if (sdev->dspless_mode_selected)
 		return 0;
 
@@ -482,6 +488,31 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 	int i;
 
+	ops = hda_dai_get_ops(substream, cpu_dai);
+	if (!ops) {
+		dev_err(cpu_dai->dev, "DAI widget ops not set\n");
+		return -EINVAL;
+	}
+
+	sdev = widget_to_sdev(w);
+	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
+
+	/* nothing more to do if the link is already prepared */
+	if (hext_stream && hext_stream->link_prepared)
+		return 0;
+
+	/*
+	 * reset the PCMSyCM registers to handle a prepare callback when the PCM is restarted
+	 * due to xruns or after a call to snd_pcm_drain/drop()
+	 */
+	ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id,
+					     0, 0, substream->stream);
+	if (ret < 0) {
+		dev_err(cpu_dai->dev, "%s:  hdac_bus_eml_sdw_map_stream_ch failed %d\n",
+			__func__, ret);
+		return ret;
+	}
+
 	data.dai_index = (link_id << 8) | cpu_dai->id;
 	data.dai_node_id = intel_alh_id;
 	ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags);
@@ -490,10 +521,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	ops = hda_dai_get_ops(substream, cpu_dai);
-	sdev = widget_to_sdev(w);
 	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
-
 	if (!hext_stream)
 		return -ENODEV;
 
-- 
2.43.0

[PATCH AUTOSEL 6.11 08/32] soundwire: intel_ace2x: Send PDI stream number during prepare

[Sasha Levin]

From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>

[ Upstream commit c78f1e15e46ac82607eed593b22992fd08644d96 ]

In the case of a prepare callback after an xrun or when the PCM is
restarted after a call to snd_pcm_drain/snd_pcm_drop, avoid
reprogramming the SHIM registers but send the PDI stream number so that
the link DMA data can be set. This is needed for the case that the DMA
data is cleared when the PCM is stopped and restarted without being
closed.

Link: https://github.com/thesofproject/sof/issues/9502
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-4-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 drivers/soundwire/intel_ace2x.c | 19 ++++++-------------
 1 file changed, 6 insertions(+), 13 deletions(-)

diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c
index 781fe0aefa68f..655766af2ea56 100644
--- a/drivers/soundwire/intel_ace2x.c
+++ b/drivers/soundwire/intel_ace2x.c
@@ -376,11 +376,12 @@ static int intel_hw_params(struct snd_pcm_substream *substream,
 static int intel_prepare(struct snd_pcm_substream *substream,
 			 struct snd_soc_dai *dai)
 {
+	struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
 	struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai);
 	struct sdw_intel *sdw = cdns_to_intel(cdns);
 	struct sdw_cdns_dai_runtime *dai_runtime;
+	struct snd_pcm_hw_params *hw_params;
 	int ch, dir;
-	int ret = 0;
 
 	dai_runtime = cdns->dai_runtime_array[dai->id];
 	if (!dai_runtime) {
@@ -389,12 +390,8 @@ static int intel_prepare(struct snd_pcm_substream *substream,
 		return -EIO;
 	}
 
+	hw_params = &rtd->dpcm[substream->stream].hw_params;
 	if (dai_runtime->suspended) {
-		struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
-		struct snd_pcm_hw_params *hw_params;
-
-		hw_params = &rtd->dpcm[substream->stream].hw_params;
-
 		dai_runtime->suspended = false;
 
 		/*
@@ -415,15 +412,11 @@ static int intel_prepare(struct snd_pcm_substream *substream,
 		/* the SHIM will be configured in the callback functions */
 
 		sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi);
-
-		/* Inform DSP about PDI stream number */
-		ret = intel_params_stream(sdw, substream, dai,
-					  hw_params,
-					  sdw->instance,
-					  dai_runtime->pdi->intel_alh_id);
 	}
 
-	return ret;
+	/* Inform DSP about PDI stream number */
+	return intel_params_stream(sdw, substream, dai, hw_params, sdw->instance,
+				   dai_runtime->pdi->intel_alh_id);
 }
 
 static int
-- 
2.43.0

[PATCH AUTOSEL 6.11 09/32] ASoC: SOF: Intel: hda: Always clean up link DMA during stop

[Sasha Levin]

From: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>

[ Upstream commit ab5593793e9088abcddce30ba8e376e31b7285fd ]

This is required to reset the DMA read/write pointers when the stream is
prepared and restarted after a call to snd_pcm_drain()/snd_pcm_drop().
Also, now that the stream is reset during stop, do not save LLP registers
in the case of STOP/suspend to avoid erroneous delay reporting.

Link: https://github.com/thesofproject/sof/issues/9502
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
All: stable@vger.kernel.org # 6.10.x 6.11.x
Link: https://patch.msgid.link/20241016032910.14601-5-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/hda-dai-ops.c | 23 ++++++++++-------------
 sound/soc/sof/intel/hda-dai.c     |  1 +
 2 files changed, 11 insertions(+), 13 deletions(-)

diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c
index 484c761478853..92681ca7f24de 100644
--- a/sound/soc/sof/intel/hda-dai-ops.c
+++ b/sound/soc/sof/intel/hda-dai-ops.c
@@ -346,20 +346,21 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_hdac_ext_stream_start(hext_stream);
 		break;
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		snd_hdac_ext_stream_clear(hext_stream);
-
 		/*
-		 * Save the LLP registers in case the stream is
-		 * restarting due PAUSE_RELEASE, or START without a pcm
-		 * close/open since in this case the LLP register is not reset
-		 * to 0 and the delay calculation will return with invalid
-		 * results.
+		 * Save the LLP registers since in case of PAUSE the LLP
+		 * register are not reset to 0, the delay calculation will use
+		 * the saved offsets for compensating the delay calculation.
 		 */
 		hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL);
 		hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU);
+		snd_hdac_ext_stream_clear(hext_stream);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+		hext_stream->pplcllpl = 0;
+		hext_stream->pplcllpu = 0;
+		snd_hdac_ext_stream_clear(hext_stream);
 		break;
 	default:
 		dev_err(sdev->dev, "unknown trigger command %d\n", cmd);
@@ -512,7 +513,6 @@ static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = {
 static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai,
 				 struct snd_pcm_substream *substream, int cmd)
 {
-	struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream);
 	struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream);
 
 	switch (cmd) {
@@ -527,9 +527,6 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c
 		if (ret < 0)
 			return ret;
 
-		if (cmd == SNDRV_PCM_TRIGGER_STOP)
-			return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
-
 		break;
 	}
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 8cccf38967e72..ac505c7ad3429 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -302,6 +302,7 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i
 	}
 
 	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		ret = hda_link_dma_cleanup(substream, hext_stream, dai);
 		if (ret < 0) {
-- 
2.43.0

[PATCH AUTOSEL 6.11 10/32] ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA

[Sasha Levin]

From: Ilya Dudikov <ilyadud@mail.ru>

[ Upstream commit b0867999e3282378a0b26a7ad200233044d31eca ]

ASUS Vivobook E1404FA needs a quirks-table entry for the internal microphone to function properly.

Signed-off-by: Ilya Dudikov <ilyadud@mail.ru>
Link: https://patch.msgid.link/20241016034038.13481-1-ilyadud25@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/amd/yc/acp6x-mach.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index e353542dcf636..ffefe45c464d0 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -325,6 +325,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "E1404FA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
-- 
2.43.0

[PATCH AUTOSEL 6.11 11/32] ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops

[Sasha Levin]

From: Derek Fang <derek.fang@realtek.com>

[ Upstream commit 6924565a04e5f424c95e6d894584e3059f257373 ]

Add a new match table entry on Lunarlake for the TM2 laptops
with rt713 and rt1318.

Signed-off-by: Derek Fang <derek.fang@realtek.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20241016030703.13669-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 .../intel/common/soc-acpi-intel-lnl-match.c   | 38 +++++++++++++++++++
 1 file changed, 38 insertions(+)

diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
index edfb668d0580d..8452b66149119 100644
--- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c
@@ -166,6 +166,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = {
 	}
 };
 
+static const struct snd_soc_acpi_adr_device rt1318_1_adr[] = {
+	{
+		.adr = 0x000133025D131801ull,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+		.name_prefix = "rt1318-1"
+	}
+};
+
 static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = {
 	{
 		.adr = 0x000130025D131801ull,
@@ -184,6 +193,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = {
 	}
 };
 
+static const struct snd_soc_acpi_adr_device rt713_0_adr[] = {
+	{
+		.adr = 0x000031025D071301ull,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+		.name_prefix = "rt713"
+	}
+};
+
 static const struct snd_soc_acpi_adr_device rt714_0_adr[] = {
 	{
 		.adr = 0x000030025D071401ull,
@@ -286,6 +304,20 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = {
 	{}
 };
 
+static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = {
+	{
+		.mask = BIT(0),
+		.num_adr = ARRAY_SIZE(rt713_0_adr),
+		.adr_d = rt713_0_adr,
+	},
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(rt1318_1_adr),
+		.adr_d = rt1318_1_adr,
+	},
+	{}
+};
+
 /* this table is used when there is no I2S codec present */
 struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
 	/* mockup tests need to be first */
@@ -343,6 +375,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg"
 	},
+	{
+		.link_mask = BIT(0) | BIT(1),
+		.links = lnl_sdw_rt713_l0_rt1318_l1,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg"
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines);
-- 
2.43.0

[PATCH AUTOSEL 6.11 18/32] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs

[Sasha Levin]

From: Alexey Klimov <alexey.klimov@linaro.org>

[ Upstream commit 9fc9ef05727ccb45fd881770f2aa5c3774b2e8e2 ]

Turns out some registers of pre-2.5 version of rxmacro codecs are not
located at the expected offsets but 0xc further away in memory. So far
the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and
CDC_RX_RX2_RX_PATH_DSM_CTL.

CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address
0x540 for RX2 but it should be 0x54C and it also overwrites
CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540.
The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n).

Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version
and above this offset will be equal to 0.
With such change the corresponding RXn() macros will generate the same
values for 2.5 codec version for all RX paths and the same old values
for pre-2.5 version for RX0 and RX1. However for the latter case with
RX2 path it will also add rxn_reg_stride2 on top.

While at this, also remove specific if-check for INTERP_AUX from
rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check
was used to handle such special offset for AUX interpolator but since
CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)
macros will generate the correst addresses of dsm register, they are no
longer needed.

Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Link: https://patch.msgid.link/20241016221049.1145101-1-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/lpass-rx-macro.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index ce42749660c87..541febab9451a 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -202,12 +202,14 @@
 #define CDC_RX_RXn_RX_PATH_SEC3(rx, n)	(0x042c  + rx->rxn_reg_stride * n)
 #define CDC_RX_RX0_RX_PATH_SEC4		(0x0430)
 #define CDC_RX_RX0_RX_PATH_SEC7		(0x0434)
-#define CDC_RX_RXn_RX_PATH_SEC7(rx, n)	(0x0434  + rx->rxn_reg_stride * n)
+#define CDC_RX_RXn_RX_PATH_SEC7(rx, n)		\
+	(0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0))
 #define CDC_RX_DSM_OUT_DELAY_SEL_MASK	GENMASK(2, 0)
 #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE	0x2
 #define CDC_RX_RX0_RX_PATH_MIX_SEC0	(0x0438)
 #define CDC_RX_RX0_RX_PATH_MIX_SEC1	(0x043C)
-#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)	(0x0440  + rx->rxn_reg_stride * n)
+#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n)	\
+	(0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0))
 #define CDC_RX_RXn_DSM_CLK_EN_MASK	BIT(0)
 #define CDC_RX_RX0_RX_PATH_DSM_CTL	(0x0440)
 #define CDC_RX_RX0_RX_PATH_DSM_DATA1	(0x0444)
@@ -645,6 +647,7 @@ struct rx_macro {
 	int rx_mclk_cnt;
 	enum lpass_codec_version codec_version;
 	int rxn_reg_stride;
+	int rxn_reg_stride2;
 	bool is_ear_mode_on;
 	bool hph_pwr_mode;
 	bool hph_hd2_mode;
@@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 							      CDC_RX_PATH_PGA_MUTE_MASK, 0x0);
 			}
 
-			if (j == INTERP_AUX)
-				dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2);
-
 			int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8;
 			int_mux_cfg1 = int_mux_cfg0 + 4;
 			int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0);
@@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component,
 
 	main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx);
 	dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx);
-	if (interp_idx == INTERP_AUX)
-		dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2);
-
 	rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx);
 
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
@@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev)
 	case LPASS_CODEC_VERSION_2_0:
 	case LPASS_CODEC_VERSION_2_1:
 		rx->rxn_reg_stride = 0x80;
+		rx->rxn_reg_stride2 = 0xc;
 		def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults);
 		reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL);
 		if (!reg_defaults)
@@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev)
 	case LPASS_CODEC_VERSION_2_7:
 	case LPASS_CODEC_VERSION_2_8:
 		rx->rxn_reg_stride = 0xc0;
+		rx->rxn_reg_stride2 = 0x0;
 		def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults);
 		reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL);
 		if (!reg_defaults)
-- 
2.43.0

[PATCH AUTOSEL 6.11 21/32] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue

[Sasha Levin]

From: Jack Yu <jack.yu@realtek.com>

[ Upstream commit 038fa6ddf5d22694f61ff7a7a53c8887c6b08c45 ]

clk_stop_timeout should be increased to 900ms to fix clock stop issue.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/cd26275d9fc54374a18dc016755cb72d@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/rt722-sdca-sdw.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index 87354bb1564e8..d5c985ff5ac55 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave)
 	}
 
 	/* set the timeout values */
-	prop->clk_stop_timeout = 200;
+	prop->clk_stop_timeout = 900;
 
 	/* wake-up event */
 	prop->wake_capable = 1;
-- 
2.43.0

[PATCH AUTOSEL 6.11 24/32] ASoC: fsl_micfil: Add sample rate constraint

[Sasha Levin]

From: Shengjiu Wang <shengjiu.wang@nxp.com>

[ Upstream commit b9a8ecf81066e01e8a3de35517481bc5aa0439e5 ]

On some platforms, for example i.MX93, there is only one
audio PLL source, so some sample rate can't be supported.
If the PLL source is used for 8kHz series rates, then 11kHz
series rates can't be supported.

So add constraints according to the frequency of available
clock sources, then alsa-lib will help to convert the
unsupported rate for the driver.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/1728884313-6778-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/fsl/fsl_micfil.c | 38 ++++++++++++++++++++++++++++++++++++++
 1 file changed, 38 insertions(+)

diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 22b240a70ad48..49afc59ed4ed9 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -28,6 +28,13 @@
 
 #define MICFIL_OSR_DEFAULT	16
 
+#define MICFIL_NUM_RATES	7
+#define MICFIL_CLK_SRC_NUM	3
+/* clock source ids */
+#define MICFIL_AUDIO_PLL1	0
+#define MICFIL_AUDIO_PLL2	1
+#define MICFIL_CLK_EXT3		2
+
 enum quality {
 	QUALITY_HIGH,
 	QUALITY_MEDIUM,
@@ -45,9 +52,12 @@ struct fsl_micfil {
 	struct clk *mclk;
 	struct clk *pll8k_clk;
 	struct clk *pll11k_clk;
+	struct clk *clk_src[MICFIL_CLK_SRC_NUM];
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct sdma_peripheral_config sdmacfg;
 	struct snd_soc_card *card;
+	struct snd_pcm_hw_constraint_list constraint_rates;
+	unsigned int constraint_rates_list[MICFIL_NUM_RATES];
 	unsigned int dataline;
 	char name[32];
 	int irq[MICFIL_IRQ_LINES];
@@ -449,12 +459,34 @@ static int fsl_micfil_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct fsl_micfil *micfil = snd_soc_dai_get_drvdata(dai);
+	unsigned int rates[MICFIL_NUM_RATES] = {8000, 11025, 16000, 22050, 32000, 44100, 48000};
+	int i, j, k = 0;
+	u64 clk_rate;
 
 	if (!micfil) {
 		dev_err(dai->dev, "micfil dai priv_data not set\n");
 		return -EINVAL;
 	}
 
+	micfil->constraint_rates.list = micfil->constraint_rates_list;
+	micfil->constraint_rates.count = 0;
+
+	for (j = 0; j < MICFIL_NUM_RATES; j++) {
+		for (i = 0; i < MICFIL_CLK_SRC_NUM; i++) {
+			clk_rate = clk_get_rate(micfil->clk_src[i]);
+			if (clk_rate != 0 && do_div(clk_rate, rates[j]) == 0) {
+				micfil->constraint_rates_list[k++] = rates[j];
+				micfil->constraint_rates.count++;
+				break;
+			}
+		}
+	}
+
+	if (micfil->constraint_rates.count > 0)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   &micfil->constraint_rates);
+
 	return 0;
 }
 
@@ -1134,6 +1166,12 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	fsl_asoc_get_pll_clocks(&pdev->dev, &micfil->pll8k_clk,
 				&micfil->pll11k_clk);
 
+	micfil->clk_src[MICFIL_AUDIO_PLL1] = micfil->pll8k_clk;
+	micfil->clk_src[MICFIL_AUDIO_PLL2] = micfil->pll11k_clk;
+	micfil->clk_src[MICFIL_CLK_EXT3] = devm_clk_get(&pdev->dev, "clkext3");
+	if (IS_ERR(micfil->clk_src[MICFIL_CLK_EXT3]))
+		micfil->clk_src[MICFIL_CLK_EXT3] = NULL;
+
 	/* init regmap */
 	regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res);
 	if (IS_ERR(regs))
-- 
2.43.0

Re: [PATCH AUTOSEL 6.11 05/32] ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 382 bytes --]

On Mon, Oct 28, 2024 at 06:49:47AM -0400, Sasha Levin wrote:
> From: Shengjiu Wang <shengjiu.wang@nxp.com>
> 
> [ Upstream commit 54c805c1eb264c839fa3027d0073bb7f323b0722 ]
> 
> Irq handler need to be executed as fast as possible, so
> the log in irq handler is better to use dev_dbg which needs
> to be enabled when debugging.

This is very marginal for stable material.

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