From 2e9261761b35f0b67b7487688cd1365f535be0b3 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Date: Tue, 16 Jun 2026 18:02:57 +0100
Subject: [PATCH 01/19] ASoC: qcom: q6apm: fix NULL pointer dereference in
 graph_callback

When q6apm_free_fragments() is called it frees rx_data.buf/tx_data.buf
and sets them to NULL under graph->lock. A late DSP buffer-done response
can race with this: graph_callback() passes the !graph->ar_graph guard
(not yet NULL), acquires the lock, but then dereferences a now-NULL buf
pointer to read buf[token].phys, crashing at virtual address 0x10.

Add a NULL check for buf inside the mutex-protected section in both the
write-done (DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2) and
read-done (DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2) handlers and bail
out cleanly if buffers have already been freed.

This problem is only shown up recently while apr bus was updated to
process the commands per service rather from single global queue.

Fixes: 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support")
Cc: Stable@vger.kernel.org
Assisted-by: Claude:claude-4-6-sonnet
Reported-by: Val Packett <val@packett.cool>
Closes: https://lore.kernel.org/all/133ced18-1aa9-475d-80d8-6120678bdde4@packett.cool/
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Link: https://patch.msgid.link/20260616170257.9381-1-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6apm.c | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 2e5b25b8d00f..641d6d243229 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -587,6 +587,10 @@ static int graph_callback(const struct gpr_resp_pkt *data, void *priv, int op)
 		token = hdr->token & APM_WRITE_TOKEN_MASK;
 
 		done = data->payload;
+		if (!graph->rx_data.buf) {
+			mutex_unlock(&graph->lock);
+			break;
+		}
 		phys = graph->rx_data.buf[token].phys;
 		mutex_unlock(&graph->lock);
 		/* token numbering starts at 0 */
@@ -609,6 +613,10 @@ static int graph_callback(const struct gpr_resp_pkt *data, void *priv, int op)
 		client_event = APM_CLIENT_EVENT_DATA_READ_DONE;
 		mutex_lock(&graph->lock);
 		rd_done = data->payload;
+		if (!graph->tx_data.buf) {
+			mutex_unlock(&graph->lock);
+			break;
+		}
 		phys = graph->tx_data.buf[hdr->token].phys;
 		mutex_unlock(&graph->lock);
 		/* token numbering starts at 0 */

From 1a3c8e28959790ae3e06029681418278b7821a3c Mon Sep 17 00:00:00 2001
From: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Date: Wed, 17 Jun 2026 13:46:04 +0200
Subject: [PATCH 02/19] ASoC: rockchip: Drop problematic guard() changes

This reverts commit f7fe9f707360 ("ASoC: rockchip: rockchip_sai: Use
guard() for spin locks").

This is very noisy pointless churn that was not tested by the submitter,
nor was it addressed to the driver's maintainer. It mixes unrelated
whitespace changes (eliminating the blank line between the includes -
why?) with hard to review diffs that add a whole indentation level to
the function for no benefit, while also not following kernel code style
by doing stuff like "ret == 0".

The driver is better off without these changes, and they're not worth
the time to validate whether they really do make no functional changes.

Signed-off-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Link: https://patch.msgid.link/20260617-sai-revert-v1-1-e46adda2213b@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_sai.c | 274 +++++++++++++++---------------
 1 file changed, 139 insertions(+), 135 deletions(-)

diff --git a/sound/soc/rockchip/rockchip_sai.c b/sound/soc/rockchip/rockchip_sai.c
index a195e96fed0a..ed393e5034a4 100644
--- a/sound/soc/rockchip/rockchip_sai.c
+++ b/sound/soc/rockchip/rockchip_sai.c
@@ -18,6 +18,7 @@
 #include <sound/pcm_params.h>
 #include <sound/dmaengine_pcm.h>
 #include <sound/tlv.h>
+
 #include "rockchip_sai.h"
 
 #define DRV_NAME		"rockchip-sai"
@@ -215,12 +216,14 @@ static void rockchip_sai_xfer_clk_stop_and_wait(struct rk_sai_dev *sai, unsigned
 static int rockchip_sai_runtime_suspend(struct device *dev)
 {
 	struct rk_sai_dev *sai = dev_get_drvdata(dev);
+	unsigned long flags;
 
 	rockchip_sai_fsync_lost_detect(sai, 0);
 	rockchip_sai_fsync_err_detect(sai, 0);
 
-	scoped_guard(spinlock_irqsave, &sai->xfer_lock)
-		rockchip_sai_xfer_clk_stop_and_wait(sai, NULL);
+	spin_lock_irqsave(&sai->xfer_lock, flags);
+	rockchip_sai_xfer_clk_stop_and_wait(sai, NULL);
+	spin_unlock_irqrestore(&sai->xfer_lock, flags);
 
 	regcache_cache_only(sai->regmap, true);
 	/*
@@ -480,6 +483,7 @@ static int rockchip_sai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai);
 	unsigned int mask = 0, val = 0;
 	unsigned int clk_gates;
+	unsigned long flags;
 	int ret = 0;
 
 	pm_runtime_get_sync(dai->dev);
@@ -495,56 +499,56 @@ static int rockchip_sai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		sai->is_master_mode = false;
 		break;
 	default:
-		pm_runtime_put(dai->dev);
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_pm_put;
 	}
 
-	scoped_guard(spinlock_irqsave, &sai->xfer_lock) {
-		rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates);
-		if (sai->initialized) {
-			if (sai->has_capture && sai->has_playback)
-				rockchip_sai_xfer_stop(sai, -1);
-			else if (sai->has_capture)
-				rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_CAPTURE);
-			else
-				rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_PLAYBACK);
-		} else {
-			rockchip_sai_clear(sai, 0);
-			sai->initialized = true;
-		}
-
-		regmap_update_bits(sai->regmap, SAI_CKR, mask, val);
-
-		mask = SAI_CKR_CKP_MASK | SAI_CKR_FSP_MASK;
-		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-		case SND_SOC_DAIFMT_NB_NF:
-			val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_NORMAL;
-			break;
-		case SND_SOC_DAIFMT_NB_IF:
-			val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_INVERTED;
-			break;
-		case SND_SOC_DAIFMT_IB_NF:
-			val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_NORMAL;
-			break;
-		case SND_SOC_DAIFMT_IB_IF:
-			val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_INVERTED;
-			break;
-		default:
-			ret = -EINVAL;
-			break;
-		}
-
-		if (ret == 0) {
-			regmap_update_bits(sai->regmap, SAI_CKR, mask, val);
-			rockchip_sai_fmt_create(sai, fmt);
-		}
-
-		if (clk_gates)
-			regmap_update_bits(sai->regmap, SAI_XFER,
-					SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK,
-					clk_gates);
+	spin_lock_irqsave(&sai->xfer_lock, flags);
+	rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates);
+	if (sai->initialized) {
+		if (sai->has_capture && sai->has_playback)
+			rockchip_sai_xfer_stop(sai, -1);
+		else if (sai->has_capture)
+			rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_CAPTURE);
+		else
+			rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_PLAYBACK);
+	} else {
+		rockchip_sai_clear(sai, 0);
+		sai->initialized = true;
 	}
 
+	regmap_update_bits(sai->regmap, SAI_CKR, mask, val);
+
+	mask = SAI_CKR_CKP_MASK | SAI_CKR_FSP_MASK;
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_NORMAL;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_INVERTED;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_NORMAL;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_INVERTED;
+		break;
+	default:
+		ret = -EINVAL;
+		goto err_xfer_unlock;
+	}
+
+	regmap_update_bits(sai->regmap, SAI_CKR, mask, val);
+
+	rockchip_sai_fmt_create(sai, fmt);
+
+err_xfer_unlock:
+	if (clk_gates)
+		regmap_update_bits(sai->regmap, SAI_XFER,
+				SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK,
+				clk_gates);
+	spin_unlock_irqrestore(&sai->xfer_lock, flags);
+err_pm_put:
 	pm_runtime_put(dai->dev);
 
 	return ret;
@@ -560,6 +564,7 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream,
 	unsigned int ch_per_lane, slot_width;
 	unsigned int val, fscr, reg;
 	unsigned int lanes, req_lanes;
+	unsigned long flags;
 	int ret = 0;
 
 	if (!rockchip_sai_stream_valid(substream, dai))
@@ -586,8 +591,8 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream,
 			dev_err(sai->dev, "not enough lanes (%d) for requested number of %s channels (%d)\n",
 				lanes, reg == SAI_TXCR ? "playback" : "capture",
 				params_channels(params));
-			pm_runtime_put(sai->dev);
-			return -EINVAL;
+			ret = -EINVAL;
+			goto err_pm_put;
 		} else {
 			lanes = req_lanes;
 		}
@@ -613,88 +618,84 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream,
 		val = SAI_XCR_VDW(32);
 		break;
 	default:
-		pm_runtime_put(sai->dev);
-		return -EINVAL;
+		ret = -EINVAL;
+		goto err_pm_put;
 	}
 
 	val |= SAI_XCR_CSR(lanes);
 
-	scoped_guard(spinlock_irqsave, &sai->xfer_lock) {
+	spin_lock_irqsave(&sai->xfer_lock, flags);
 
-		regmap_update_bits(sai->regmap, reg, SAI_XCR_VDW_MASK | SAI_XCR_CSR_MASK, val);
+	regmap_update_bits(sai->regmap, reg, SAI_XCR_VDW_MASK | SAI_XCR_CSR_MASK, val);
 
-		if (!sai->is_tdm)
-			regmap_update_bits(sai->regmap, reg, SAI_XCR_SBW_MASK,
-					   SAI_XCR_SBW(params_physical_width(params)));
+	if (!sai->is_tdm)
+		regmap_update_bits(sai->regmap, reg, SAI_XCR_SBW_MASK,
+				   SAI_XCR_SBW(params_physical_width(params)));
 
-		regmap_read(sai->regmap, reg, &val);
+	regmap_read(sai->regmap, reg, &val);
 
-		slot_width = SAI_XCR_SBW_V(val);
-		ch_per_lane = params_channels(params) / lanes;
+	slot_width = SAI_XCR_SBW_V(val);
+	ch_per_lane = params_channels(params) / lanes;
 
-		regmap_update_bits(sai->regmap, reg, SAI_XCR_SNB_MASK,
-				   SAI_XCR_SNB(ch_per_lane));
+	regmap_update_bits(sai->regmap, reg, SAI_XCR_SNB_MASK,
+			   SAI_XCR_SNB(ch_per_lane));
 
-		fscr = SAI_FSCR_FW(sai->fw_ratio * slot_width * ch_per_lane);
+	fscr = SAI_FSCR_FW(sai->fw_ratio * slot_width * ch_per_lane);
 
-		switch (sai->fpw) {
-		case FPW_ONE_BCLK_WIDTH:
-			fscr |= SAI_FSCR_FPW(1);
-			break;
-		case FPW_ONE_SLOT_WIDTH:
-			fscr |= SAI_FSCR_FPW(slot_width);
-			break;
-		case FPW_HALF_FRAME_WIDTH:
-			fscr |= SAI_FSCR_FPW(sai->fw_ratio * slot_width * ch_per_lane / 2);
-			break;
-		default:
-			dev_err(sai->dev, "Invalid Frame Pulse Width %d\n", sai->fpw);
-			ret = -EINVAL;
-			break;
-		}
-
-		if (ret == 0) {
-			regmap_update_bits(sai->regmap, SAI_FSCR,
-					   SAI_FSCR_FW_MASK | SAI_FSCR_FPW_MASK, fscr);
-
-			if (sai->is_master_mode) {
-				bclk_rate = sai->fw_ratio * slot_width *
-						ch_per_lane * params_rate(params);
-				ret = clk_set_rate(sai->mclk, sai->mclk_rate);
-				if (ret)
-					dev_err(sai->dev, "Failed to set mclk to %u: %pe\n",
-						sai->mclk_rate, ERR_PTR(ret));
-				else {
-					mclk_rate = clk_get_rate(sai->mclk);
-					if (mclk_rate < bclk_rate) {
-						dev_err(sai->dev, "Mismatch mclk: %u, at least %u\n",
-							mclk_rate, bclk_rate);
-						ret = -EINVAL;
-					} else {
-
-						div_bclk = DIV_ROUND_CLOSEST(mclk_rate, bclk_rate);
-						mclk_req_rate = bclk_rate * div_bclk;
-
-						if (mclk_rate <
-							mclk_req_rate - CLK_SHIFT_RATE_HZ_MAX ||
-						    mclk_rate >
-							mclk_req_rate + CLK_SHIFT_RATE_HZ_MAX) {
-							dev_err(sai->dev,
-								"Mismatch mclk: %u, expected %u (+/- %dHz)\n",
-								mclk_rate, mclk_req_rate,
-								CLK_SHIFT_RATE_HZ_MAX);
-							ret = -EINVAL;
-						} else
-							regmap_update_bits(sai->regmap,
-									SAI_CKR,
-									SAI_CKR_MDIV_MASK,
-									SAI_CKR_MDIV(div_bclk));
-					}
-				}
-			}
-		}
+	switch (sai->fpw) {
+	case FPW_ONE_BCLK_WIDTH:
+		fscr |= SAI_FSCR_FPW(1);
+		break;
+	case FPW_ONE_SLOT_WIDTH:
+		fscr |= SAI_FSCR_FPW(slot_width);
+		break;
+	case FPW_HALF_FRAME_WIDTH:
+		fscr |= SAI_FSCR_FPW(sai->fw_ratio * slot_width * ch_per_lane / 2);
+		break;
+	default:
+		dev_err(sai->dev, "Invalid Frame Pulse Width %d\n", sai->fpw);
+		ret = -EINVAL;
+		goto err_xfer_unlock;
 	}
 
+	regmap_update_bits(sai->regmap, SAI_FSCR,
+			   SAI_FSCR_FW_MASK | SAI_FSCR_FPW_MASK, fscr);
+
+	if (sai->is_master_mode) {
+		bclk_rate = sai->fw_ratio * slot_width * ch_per_lane * params_rate(params);
+		ret = clk_set_rate(sai->mclk, sai->mclk_rate);
+		if (ret) {
+			dev_err(sai->dev, "Failed to set mclk to %u: %pe\n",
+				sai->mclk_rate, ERR_PTR(ret));
+			goto err_xfer_unlock;
+		}
+
+		mclk_rate = clk_get_rate(sai->mclk);
+		if (mclk_rate < bclk_rate) {
+			dev_err(sai->dev, "Mismatch mclk: %u, at least %u\n",
+				mclk_rate, bclk_rate);
+			ret = -EINVAL;
+			goto err_xfer_unlock;
+		}
+
+		div_bclk = DIV_ROUND_CLOSEST(mclk_rate, bclk_rate);
+		mclk_req_rate = bclk_rate * div_bclk;
+
+		if (mclk_rate < mclk_req_rate - CLK_SHIFT_RATE_HZ_MAX ||
+		    mclk_rate > mclk_req_rate + CLK_SHIFT_RATE_HZ_MAX) {
+			dev_err(sai->dev, "Mismatch mclk: %u, expected %u (+/- %dHz)\n",
+				mclk_rate, mclk_req_rate, CLK_SHIFT_RATE_HZ_MAX);
+			ret = -EINVAL;
+			goto err_xfer_unlock;
+		}
+
+		regmap_update_bits(sai->regmap, SAI_CKR, SAI_CKR_MDIV_MASK,
+				   SAI_CKR_MDIV(div_bclk));
+	}
+
+err_xfer_unlock:
+	spin_unlock_irqrestore(&sai->xfer_lock, flags);
+err_pm_put:
 	pm_runtime_put(sai->dev);
 
 	return ret;
@@ -704,6 +705,7 @@ static int rockchip_sai_prepare(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai);
+	unsigned long flags;
 
 	if (!rockchip_sai_stream_valid(substream, dai))
 		return 0;
@@ -724,12 +726,13 @@ static int rockchip_sai_prepare(struct snd_pcm_substream *substream,
 		 * udelay falls short.
 		 */
 		udelay(20);
-		scoped_guard(spinlock_irqsave, &sai->xfer_lock)
-			regmap_update_bits(sai->regmap, SAI_XFER,
-					   SAI_XFER_CLK_MASK |
-					   SAI_XFER_FSS_MASK,
-					   SAI_XFER_CLK_EN |
-					   SAI_XFER_FSS_EN);
+		spin_lock_irqsave(&sai->xfer_lock, flags);
+		regmap_update_bits(sai->regmap, SAI_XFER,
+				   SAI_XFER_CLK_MASK |
+				   SAI_XFER_FSS_MASK,
+				   SAI_XFER_CLK_EN |
+				   SAI_XFER_FSS_EN);
+		spin_unlock_irqrestore(&sai->xfer_lock, flags);
 	}
 
 	rockchip_sai_fsync_lost_detect(sai, 1);
@@ -912,6 +915,7 @@ static int rockchip_sai_set_tdm_slot(struct snd_soc_dai *dai,
 				     int slots, int slot_width)
 {
 	struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai);
+	unsigned long flags;
 	unsigned int clk_gates;
 	int sw = slot_width;
 
@@ -927,16 +931,16 @@ static int rockchip_sai_set_tdm_slot(struct snd_soc_dai *dai,
 		return -EINVAL;
 
 	pm_runtime_get_sync(dai->dev);
-	scoped_guard(spinlock_irqsave, &sai->xfer_lock) {
-		rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates);
-		regmap_update_bits(sai->regmap, SAI_TXCR, SAI_XCR_SBW_MASK,
-				   SAI_XCR_SBW(sw));
-		regmap_update_bits(sai->regmap, SAI_RXCR, SAI_XCR_SBW_MASK,
-				   SAI_XCR_SBW(sw));
-		regmap_update_bits(sai->regmap, SAI_XFER,
-				   SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK,
-				   clk_gates);
-	}
+	spin_lock_irqsave(&sai->xfer_lock, flags);
+	rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates);
+	regmap_update_bits(sai->regmap, SAI_TXCR, SAI_XCR_SBW_MASK,
+			   SAI_XCR_SBW(sw));
+	regmap_update_bits(sai->regmap, SAI_RXCR, SAI_XCR_SBW_MASK,
+			   SAI_XCR_SBW(sw));
+	regmap_update_bits(sai->regmap, SAI_XFER,
+			   SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK,
+			   clk_gates);
+	spin_unlock_irqrestore(&sai->xfer_lock, flags);
 	pm_runtime_put(dai->dev);
 
 	return 0;

From fdf043f5f3bae150b678feae3d7bb1beed87ec14 Mon Sep 17 00:00:00 2001
From: Sen Wang <sen@ti.com>
Date: Tue, 16 Jun 2026 18:33:22 -0500
Subject: [PATCH 03/19] ASoC: tlv320aic3x: restrict CLKDIV bypass Q values in
 dual-rate mode

The datasheet documents that when the PLL is disabled and dual-rate mode
is enabled, only Q values {4, 8, 9, 12, 16} are valid for the CLKDIV
bypass path; all other Q values produce invalid bitclock output.

The existing loop iterates Q from 2 to 17 without this restriction,
causing silent audio failure when an out-of-spec Q is picked.

Restrict the Q search to the allowed set in dual-rate mode.

Fixes: 4f9c16ccfa26 ("[ALSA] soc - tlv320aic3x - revisit clock setup")
Suggested-by: Mir Jeffres <m-jeffres@ti.com>
Signed-off-by: Sen Wang <sen@ti.com>
Link: https://patch.msgid.link/20260616233322.873081-1-sen@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++------
 1 file changed, 19 insertions(+), 6 deletions(-)

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b12c1952823b..b38393a8130f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1049,11 +1049,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
 {
+	static const u8 dual_rate_q[] = {4, 8, 9, 12, 16};
 	struct snd_soc_component *component = dai->component;
 	struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 d, pll_d = 1;
+	bool dual_rate;
 	int clk;
 	int width = aic3x->slot_width;
 
@@ -1079,14 +1081,25 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 
 	/* Fsref can be 44100 or 48000 */
 	fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
+	dual_rate = params_rate(params) >= 64000;
 
 	/* Try to find a value for Q which allows us to bypass the PLL and
 	 * generate CODEC_CLK directly. */
-	for (pll_q = 2; pll_q < 18; pll_q++)
-		if (aic3x->sysclk / (128 * pll_q) == fsref) {
-			bypass_pll = 1;
-			break;
+	if (dual_rate) {
+		for (int i = 0; i < ARRAY_SIZE(dual_rate_q); i++) {
+			pll_q = dual_rate_q[i];
+			if (aic3x->sysclk / (128 * pll_q) == fsref) {
+				bypass_pll = 1;
+				break;
+			}
 		}
+	} else {
+		for (pll_q = 2; pll_q < 18; pll_q++)
+			if (aic3x->sysclk / (128 * pll_q) == fsref) {
+				bypass_pll = 1;
+				break;
+			}
+	}
 
 	if (bypass_pll) {
 		pll_q &= 0xf;
@@ -1106,13 +1119,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	 * right DAC to right channel input */
 	data = (LDAC2LCH | RDAC2RCH);
 	data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
-	if (params_rate(params) >= 64000)
+	if (dual_rate)
 		data |= DUAL_RATE_MODE;
 	snd_soc_component_write(component, AIC3X_CODEC_DATAPATH_REG, data);
 
 	/* codec sample rate select */
 	data = (fsref * 20) / params_rate(params);
-	if (params_rate(params) < 64000)
+	if (!dual_rate)
 		data /= 2;
 	data /= 5;
 	data -= 2;

From 5096a1634def431cd7f89e2b0af4456de91dd26d Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Mon, 15 Jun 2026 17:10:12 +0800
Subject: [PATCH 04/19] ASoC: rt5650: enhance spk protection function

This patch adjusts several default settings to ensure
the speaker protection function can be enabled safely.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://patch.msgid.link/20260615091012.718168-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5645.c | 6 ++----
 sound/soc/codecs/rt5645.h | 1 +
 2 files changed, 3 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 8a9af260e5f7..e9819653b30d 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -83,6 +83,8 @@ static const struct reg_sequence rt5650_init_list[] = {
 	{RT5645_PWR_ANLG1, 0x02},
 	{RT5645_IL_CMD3, 0x6728},
 	{RT5645_PR_BASE + 0x3a,	0x0000},
+	{RT5645_CLSD_OUT_CTRL1, 0x4059},
+	{RT5645_GEN_CTRL3, 0x0200},
 };
 
 static const struct reg_default rt5645_reg[] = {
@@ -1855,13 +1857,9 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w,
 			RT5645_PWR_CLS_D_L,
 			RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
 			RT5645_PWR_CLS_D_L);
-		snd_soc_component_update_bits(component, RT5645_GEN_CTRL3,
-			RT5645_DET_CLK_MASK, RT5645_DET_CLK_MODE1);
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		snd_soc_component_update_bits(component, RT5645_GEN_CTRL3,
-			RT5645_DET_CLK_MASK, RT5645_DET_CLK_DIS);
 		snd_soc_component_write(component, RT5645_EQ_CTRL2, 0);
 		snd_soc_component_update_bits(component, RT5645_PWR_DIG1,
 			RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R |
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index bef74b29fd54..a5bfe9861b8b 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -118,6 +118,7 @@
 #define RT5645_A_JD_CTRL1			0x94
 #define RT5645_VAD_CTRL4			0x9d
 #define RT5645_CLSD_OUT_CTRL			0xa0
+#define RT5645_CLSD_OUT_CTRL1			0xa1
 /* Function - Digital */
 #define RT5645_ADC_EQ_CTRL1			0xae
 #define RT5645_ADC_EQ_CTRL2			0xaf

From e6fa716c9d7248e03cb93f566874bd5709901bcf Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= <alvin.sipraga@analog.com>
Date: Wed, 17 Jun 2026 16:55:08 +0200
Subject: [PATCH 05/19] ASoC: audio-graph-card2: Drop warning for manually
 selected DAI formats
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Sometimes the DAI format must be specified in the audio-graph-card2
device tree, so emitting a warning can be misleading. Revert back to
emitting no warning.

A few examples where automatic format selection might not be applicable:

- For DPCM, where the other side of the DAI link is not apparent, no
  proper selection can actually be made. This can lead to disagreeing
  formats.

- Due to hardware peculiarities, some ostensibly supported formats might
  not work in practice.

In either case, the only correct solution is for the sound card to set
the format

Link: https://lore.kernel.org/all/87ik7s36k2.wl-kuninori.morimoto.gx@renesas.com/
Signed-off-by: Alvin Šipraga <alvin.sipraga@analog.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/20260617145508.327213-1-alvin@pqrs.dk
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card2.c | 12 ------------
 1 file changed, 12 deletions(-)

diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index 6894bb936cfd..0202ed0ee78e 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -778,18 +778,6 @@ static void graph_link_init(struct simple_util_priv *priv,
 	graph_parse_daifmt(ports_cpu,	&daifmt);
 	graph_parse_daifmt(ports_codec,	&daifmt);
 	graph_parse_daifmt(lnk,		&daifmt);
-	if (daifmt) {
-		struct device *dev = simple_priv_to_dev(priv);
-
-		/*
-		 * Recommend to use Auto Select by using .auto_selectable_formats.
-		 * linux/sound/soc/renesas/rcar/core.c can be good sample for it.
-		 *
-		 * One note is that Audio Graph Card2 still keeps compatible to set
-		 * DAI format via DT.
-		 */
-		dev_warn_once(dev, "use .auto_selectable_formats on each corresponding CPU/Codec driver");
-	}
 
 	graph_util_parse_link_direction(lnk,		&playback_only, &capture_only);
 	graph_util_parse_link_direction(ports_cpu,	&playback_only, &capture_only);

From cf81b260916c92cf610c907d41627a175e86e37d Mon Sep 17 00:00:00 2001
From: Ahmad Fatoum <a.fatoum@pengutronix.de>
Date: Wed, 17 Jun 2026 16:47:53 +0200
Subject: [PATCH 06/19] ASoC: cs530x: Fix expected MCLK rates for CS5302/4/8

When this driver was first added, it accepted rates of 24.56 MHz and
22.572 MHz for the MCLK when PLL bypass is enabled.

These rates seem to have no basis in the datasheets and were thus replaced
with 45.1584 MHz and 49.152 MHz, respectively, in commit e7ab858390f2
("ASoC: cs530x: Correct MCLK reference frequency values").

While the new rates are indeed correct for the CS4xxx ICs[0][1][2][3],
they are incorrect for the CS530x ICs the driver was originally written to
support as the MCLK frequencies are halved there[4][5][6].

Fix this by checking against the correct type-appropriate rates.

While at it, drop the CS530X_SYSCLK_REF_* macros. They arguably confuse
more than they help, especially as they are not applicable to the
cs5302/4/8.

[0]: https://statics.cirrus.com/pubs/proDatasheet/CS4282P_DS1318F1.pdf
[1]: https://statics.cirrus.com/pubs/proDatasheet/CS4302P_DS1315F1.pdf
[2]: https://statics.cirrus.com/pubs/proDatasheet/CS4304P_DS1316F1.pdf
[3]: https://statics.cirrus.com/pubs/proDatasheet/CS4308P_DS1317F1.pdf
[4]: https://statics.cirrus.com/pubs/proDatasheet/CS5302P_DS1312F1.pdf
[5]: https://statics.cirrus.com/pubs/proDatasheet/CS5304P_DS1313F1.pdf
[6]: https://statics.cirrus.com/pubs/proDatasheet/CS5308P_DS1314F1.pdf

Fixes: 2884c29152c0 ("ASoC: cs530x: Support for cs530x ADCs")
Signed-off-by: Ahmad Fatoum <a.fatoum@pengutronix.de>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260617-cs530x-mclk-v1-1-0215b5f1a0a4@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs530x.c | 29 ++++++++++++++++++++++++-----
 sound/soc/codecs/cs530x.h |  6 ------
 2 files changed, 24 insertions(+), 11 deletions(-)

diff --git a/sound/soc/codecs/cs530x.c b/sound/soc/codecs/cs530x.c
index 18b5ff75feec..2c7e33135911 100644
--- a/sound/soc/codecs/cs530x.c
+++ b/sound/soc/codecs/cs530x.c
@@ -1093,6 +1093,29 @@ static int cs530x_component_probe(struct snd_soc_component *component)
 	return 0;
 }
 
+static bool cs530x_mclk_freq_is_valid(struct cs530x_priv *cs530x,
+				      unsigned int freq)
+{
+	/*
+	 * All these chips support 48 kHz- and 44.1 kHz-related sample rates,
+	 * but they differ in what MCLK frequency is required for achieving
+	 * the sample rate.
+	 */
+	switch (cs530x->devtype) {
+	case CS4282:
+	case CS4302:
+	case CS4304:
+	case CS4308:
+		return freq == 49152000 || freq == 45158400;
+	case CS5302:
+	case CS5304:
+	case CS5308:
+		return freq == 24576000 || freq == 22579200;
+	}
+
+	return false;
+}
+
 static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id,
 			     int source, unsigned int freq, int dir)
 {
@@ -1101,11 +1124,7 @@ static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id,
 
 	switch (source) {
 	case CS530X_SYSCLK_SRC_MCLK:
-		switch (freq) {
-		case CS530X_SYSCLK_REF_45_1MHZ:
-		case CS530X_SYSCLK_REF_49_1MHZ:
-			break;
-		default:
+		if (!cs530x_mclk_freq_is_valid(cs530x, freq)) {
 			dev_err(component->dev, "Invalid MCLK source rate %d\n", freq);
 			return -EINVAL;
 		}
diff --git a/sound/soc/codecs/cs530x.h b/sound/soc/codecs/cs530x.h
index 1e2f6a7a589c..18aa4dfd0c86 100644
--- a/sound/soc/codecs/cs530x.h
+++ b/sound/soc/codecs/cs530x.h
@@ -200,12 +200,6 @@
 /* IN_VOL_CTL5 and OUT_VOL_CTL5 */
 #define CS530X_INOUT_VU			BIT(0)
 
-/* MCLK Reference Source Frequency */
-/* 41KHz related */
-#define CS530X_SYSCLK_REF_45_1MHZ	45158400
-/* 48KHz related */
-#define CS530X_SYSCLK_REF_49_1MHZ	49152000
-
 /* System Clock Source */
 #define CS530X_SYSCLK_SRC_MCLK		0
 #define CS530X_SYSCLK_SRC_PLL		1

From 5714c8359f4fc171ab8e0bd0dfc4c61fb36e1db6 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 11:18:10 +0100
Subject: [PATCH 07/19] soundwire: Move wait for initialisation helper to
 header

As SoundWire devices tend to enumerate on the bus after probe, drivers
frequently need to wait for the device to initialise from common driver
code. The common system is to split drivers into a core module and then
a module for each communication bus. These two facts tend to cause
Kconfig issues, the issue tends to be when SOUNDWIRE=m and DRIVER_I2C=y,
this usually selects DRIVER=y. The driver code then wants to call
sdw_slave_wait_for_init(), but this results in calling a module function
from built in code. A depends on SOUNDWIRE | !SOUNDWIRE could be added to
the end driver but this seems slightly off as it adds a lot of counter
intuitive depends.

A simpler solution is to make sdw_slave_wait_for_init() a static inline
function. As part of doing this add a check for the slave device being
NULL acknowledging that this is likely called from code that is shared
between control buses. It does require dropping the call to
sdw_show_ping_status() but this can be added back in end drivers that
used it originally.

Currently this is causing rand config issues on RT5682 and will soon
also cause similar problems on cs42l43.

Acked-by: Vinod Koul <vkoul@kernel.org>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623101814.24044-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 drivers/soundwire/bus.c       | 28 --------------------------
 include/linux/soundwire/sdw.h | 38 +++++++++++++++++++++++++++--------
 2 files changed, 30 insertions(+), 36 deletions(-)

diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c
index b7bdf19ebb42..fe5316d93fef 100644
--- a/drivers/soundwire/bus.c
+++ b/drivers/soundwire/bus.c
@@ -1372,34 +1372,6 @@ int sdw_slave_get_current_bank(struct sdw_slave *slave)
 }
 EXPORT_SYMBOL_GPL(sdw_slave_get_current_bank);
 
-/**
- * sdw_slave_wait_for_init - Wait for device initialisation
- * @slave: Pointer to the SoundWire peripheral.
- * @timeout_ms: Timeout in milliseconds.
- *
- * Wait for a peripheral device to enumerate and be initialised by the
- * SoundWire core.
- *
- * Return: Zero on success, and a negative error code on failure.
- */
-int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms)
-{
-	unsigned long time;
-
-	time = wait_for_completion_timeout(&slave->initialization_complete,
-					   msecs_to_jiffies(timeout_ms));
-	if (!time) {
-		dev_err(&slave->dev, "Initialization not complete\n");
-		sdw_show_ping_status(slave->bus, true);
-		return -ETIMEDOUT;
-	}
-
-	slave->unattach_request = 0;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(sdw_slave_wait_for_init);
-
 static int sdw_slave_set_frequency(struct sdw_slave *slave)
 {
 	int scale_index;
diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h
index a46cbaec5949..b484784e2690 100644
--- a/include/linux/soundwire/sdw.h
+++ b/include/linux/soundwire/sdw.h
@@ -11,6 +11,7 @@
 #include <linux/idr.h>
 #include <linux/irq.h>
 #include <linux/irqdomain.h>
+#include <linux/jiffies.h>
 #include <linux/lockdep_types.h>
 #include <linux/mod_devicetable.h>
 #include <linux/mutex.h>
@@ -1093,8 +1094,6 @@ int sdw_slave_get_current_bank(struct sdw_slave *sdev);
 
 int sdw_slave_get_scale_index(struct sdw_slave *slave, u8 *base);
 
-int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms);
-
 /* messaging and data APIs */
 int sdw_read(struct sdw_slave *slave, u32 addr);
 int sdw_write(struct sdw_slave *slave, u32 addr, u8 value);
@@ -1138,12 +1137,6 @@ static inline int sdw_slave_get_current_bank(struct sdw_slave *sdev)
 	return -EINVAL;
 }
 
-static inline int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms)
-{
-	WARN_ONCE(1, "SoundWire API is disabled");
-	return -EINVAL;
-}
-
 /* messaging and data APIs */
 static inline int sdw_read(struct sdw_slave *slave, u32 addr)
 {
@@ -1207,4 +1200,33 @@ static inline int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u
 
 #endif /* CONFIG_SOUNDWIRE */
 
+/**
+ * sdw_slave_wait_for_init - Wait for device initialisation
+ * @slave: Pointer to the SoundWire peripheral.
+ * @timeout_ms: Timeout in milliseconds.
+ *
+ * Wait for a peripheral device to enumerate and be initialised by the
+ * SoundWire core.
+ *
+ * Return: Zero on success, and a negative error code on failure.
+ */
+static inline int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms)
+{
+	unsigned long time;
+
+	if (!slave)
+		return 0;
+
+	time = wait_for_completion_timeout(&slave->initialization_complete,
+					   msecs_to_jiffies(timeout_ms));
+	if (!time) {
+		dev_err(&slave->dev, "Initialization not complete\n");
+		return -ETIMEDOUT;
+	}
+
+	slave->unattach_request = 0;
+
+	return 0;
+}
+
 #endif /* __SOUNDWIRE_H */

From ce52450319fb0ee122e5be5ec8dfb888ee1e0237 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 11:18:11 +0100
Subject: [PATCH 08/19] ASoC: es9356: Add back local call to
 sdw_show_ping_status()

As the core no longer calls this debug helper add it back to the drivers
that originally called it.

Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623101814.24044-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es9356.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/es9356.c b/sound/soc/codecs/es9356.c
index 670e918b56a4..8db81d574624 100644
--- a/sound/soc/codecs/es9356.c
+++ b/sound/soc/codecs/es9356.c
@@ -1111,8 +1111,10 @@ static int es9356_sdca_dev_resume(struct device *dev)
 		es9356->disable_irq = false;
 
 	ret = sdw_slave_wait_for_init(slave, es9356_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(es9356->regmap, false);
 	regcache_sync(es9356->regmap);

From 1921303a1d2f26ae70446aa18fb218767bddd913 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 11:18:12 +0100
Subject: [PATCH 09/19] ASoC: max98373: Add back local call to
 sdw_show_ping_status()

As the core no longer calls this debug helper add it back to the drivers
that originally called it.

Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623101814.24044-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98373-sdw.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index 6829fa07c9ec..7a42052dc051 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -272,8 +272,10 @@ static int max98373_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, MAX98373_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(max98373->regmap, false);
 	regcache_sync(max98373->regmap);

From ea9ff3b7bcfbcd1e61d34590c7c632005ef3d9aa Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 11:18:13 +0100
Subject: [PATCH 10/19] ASoC: ti: Add back local call to sdw_show_ping_status()

As the core no longer calls this debug helper add it back to the drivers
that originally called it.

Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623101814.24044-5-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tac5xx2-sdw.c | 4 +++-
 sound/soc/codecs/tas2783-sdw.c | 4 +++-
 2 files changed, 6 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/tac5xx2-sdw.c b/sound/soc/codecs/tac5xx2-sdw.c
index bb12cfb6da12..ace06f5ab58c 100644
--- a/sound/soc/codecs/tac5xx2-sdw.c
+++ b/sound/soc/codecs/tac5xx2-sdw.c
@@ -1445,8 +1445,10 @@ static s32 tac5xx2_sdca_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, TAC5XX2_PROBE_TIMEOUT_MS);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(tac_dev->regmap, false);
 	regcache_mark_dirty(tac_dev->regmap);
diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c
index 7d70e7e3f24f..1127ea59b5e4 100644
--- a/sound/soc/codecs/tas2783-sdw.c
+++ b/sound/soc/codecs/tas2783-sdw.c
@@ -1083,8 +1083,10 @@ static s32 tas2783_sdca_dev_resume(struct device *dev)
 	int ret;
 
 	ret = sdw_slave_wait_for_init(slave, TAS2783_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(tas_dev->regmap, false);
 	regcache_sync(tas_dev->regmap);

From 6540b9d9ccc32ad1546dcc7b4d4bcbb68c667714 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 11:18:14 +0100
Subject: [PATCH 11/19] ASoC: realtek: Add back local call to
 sdw_show_ping_status()

As the core no longer calls this debug helper add it back to the drivers
that originally called it.

Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623101814.24044-6-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1017-sdca-sdw.c | 4 +++-
 sound/soc/codecs/rt1308-sdw.c      | 4 +++-
 sound/soc/codecs/rt1316-sdw.c      | 4 +++-
 sound/soc/codecs/rt5682-sdw.c      | 4 +++-
 sound/soc/codecs/rt700-sdw.c       | 4 +++-
 sound/soc/codecs/rt711-sdca-sdw.c  | 4 +++-
 sound/soc/codecs/rt712-sdca-dmic.c | 4 +++-
 sound/soc/codecs/rt712-sdca-sdw.c  | 4 +++-
 sound/soc/codecs/rt715-sdca-sdw.c  | 4 +++-
 sound/soc/codecs/rt715-sdw.c       | 4 +++-
 sound/soc/codecs/rt721-sdca-sdw.c  | 4 +++-
 sound/soc/codecs/rt722-sdca-sdw.c  | 4 +++-
 12 files changed, 36 insertions(+), 12 deletions(-)

diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c
index d62e8a253676..91d3d43cd998 100644
--- a/sound/soc/codecs/rt1017-sdca-sdw.c
+++ b/sound/soc/codecs/rt1017-sdca-sdw.c
@@ -779,8 +779,10 @@ static int rt1017_sdca_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT1017_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt1017->regmap, false);
 	regcache_sync(rt1017->regmap);
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index 39e06a3a7560..60e5040b6dd9 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -774,8 +774,10 @@ static int rt1308_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT1308_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt1308->regmap, false);
 	regcache_sync_region(rt1308->regmap, 0xc000, 0xcfff);
diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c
index 1828fd9d5af6..5e8eda6a5f7f 100644
--- a/sound/soc/codecs/rt1316-sdw.c
+++ b/sound/soc/codecs/rt1316-sdw.c
@@ -751,8 +751,10 @@ static int rt1316_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT1316_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt1316->regmap, false);
 	regcache_sync(rt1316->regmap);
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index ec2a35a0cacd..dec8c2147d68 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -769,8 +769,10 @@ static int rt5682_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, RT5682_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt5682->sdw_regmap, false);
 	regcache_cache_only(rt5682->regmap, false);
diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c
index 30fcca210f05..6bc636c86f42 100644
--- a/sound/soc/codecs/rt700-sdw.c
+++ b/sound/soc/codecs/rt700-sdw.c
@@ -528,8 +528,10 @@ static int rt700_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT700_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt700->regmap, false);
 	regcache_sync_region(rt700->regmap, 0x3000, 0x8fff);
diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c
index a8164fc3979a..461315844ba9 100644
--- a/sound/soc/codecs/rt711-sdca-sdw.c
+++ b/sound/soc/codecs/rt711-sdca-sdw.c
@@ -454,8 +454,10 @@ static int rt711_sdca_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, RT711_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt711->regmap, false);
 	regcache_sync(rt711->regmap);
diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c
index 4c5c2f5ba5ed..8b7d50a80ff9 100644
--- a/sound/soc/codecs/rt712-sdca-dmic.c
+++ b/sound/soc/codecs/rt712-sdca-dmic.c
@@ -911,8 +911,10 @@ static int rt712_sdca_dmic_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT712_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt712->regmap, false);
 	regcache_sync(rt712->regmap);
diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c
index 581732180473..2787524c796e 100644
--- a/sound/soc/codecs/rt712-sdca-sdw.c
+++ b/sound/soc/codecs/rt712-sdca-sdw.c
@@ -467,8 +467,10 @@ static int rt712_sdca_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, RT712_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt712->regmap, false);
 	regcache_sync(rt712->regmap);
diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c
index 4b9815b5628d..fabd21bbbe5b 100644
--- a/sound/soc/codecs/rt715-sdca-sdw.c
+++ b/sound/soc/codecs/rt715-sdca-sdw.c
@@ -230,8 +230,10 @@ static int rt715_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT715_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt715->regmap, false);
 	regcache_sync_region(rt715->regmap,
diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c
index 7f83a8f1a06e..a4a3945522e8 100644
--- a/sound/soc/codecs/rt715-sdw.c
+++ b/sound/soc/codecs/rt715-sdw.c
@@ -507,8 +507,10 @@ static int rt715_dev_resume(struct device *dev)
 		return 0;
 
 	ret = sdw_slave_wait_for_init(slave, RT715_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt715->regmap, false);
 	regcache_sync_region(rt715->regmap, 0x3000, 0x8fff);
diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c
index 58606209316a..02df04a0ddad 100644
--- a/sound/soc/codecs/rt721-sdca-sdw.c
+++ b/sound/soc/codecs/rt721-sdca-sdw.c
@@ -505,8 +505,10 @@ static int rt721_sdca_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, RT721_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt721->regmap, false);
 	regcache_sync(rt721->regmap);
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index 0f76492ff915..284900933ebf 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -552,8 +552,10 @@ static int rt722_sdca_dev_resume(struct device *dev)
 	}
 
 	ret = sdw_slave_wait_for_init(slave, RT722_PROBE_TIMEOUT);
-	if (ret)
+	if (ret) {
+		sdw_show_ping_status(slave->bus, true);
 		return ret;
+	}
 
 	regcache_cache_only(rt722->regmap, false);
 	regcache_sync(rt722->regmap);

From 1ce42a11bed134903e352010a01fa53073a6b395 Mon Sep 17 00:00:00 2001
From: HyeongJun An <sammiee5311@gmail.com>
Date: Tue, 23 Jun 2026 20:05:26 +0900
Subject: [PATCH 12/19] ASoC: SDCA: Validate written enum value in
 ge_put_enum_double()

ge_put_enum_double() passes the user-supplied enumeration index
item[0] to snd_soc_enum_item_to_val() without checking it against the
number of items in the enum:

	ret = snd_soc_enum_item_to_val(e, item[0]);

snd_soc_enum_item_to_val() indexes the heap-allocated e->values[] array
with that index (e->values is set from a devm_kcalloc() of e->items
entries), so a control write with an out-of-range item[0] reads past the
end of the values buffer.  The bounds check in
snd_soc_dapm_put_enum_double() only runs afterwards, so it does not
prevent the read here.

Reject an out-of-range item before using it, matching the other enum put
handlers.

This issue was pointed out by the Sashiko AI review bot while reviewing a
related enum-validation series:
https://lore.kernel.org/all/20260609125735.CEB651F00893@smtp.kernel.org/

Fixes: 812ff1baa764 ("ASoC: SDCA: Limit values user can write to Selected Mode")
Signed-off-by: HyeongJun An <sammiee5311@gmail.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623110526.813217-1-sammiee5311@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sdca/sdca_asoc.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/sdca/sdca_asoc.c b/sound/soc/sdca/sdca_asoc.c
index e76afa396b0a..b4dedba719dc 100644
--- a/sound/soc/sdca/sdca_asoc.c
+++ b/sound/soc/sdca/sdca_asoc.c
@@ -160,6 +160,9 @@ static int ge_put_enum_double(struct snd_kcontrol *kcontrol,
 	unsigned int reg = e->reg;
 	int ret;
 
+	if (item[0] >= e->items)
+		return -EINVAL;
+
 	reg &= ~SDW_SDCA_CTL_CSEL(0x3F);
 	reg |= SDW_SDCA_CTL_CSEL(SDCA_CTL_GE_DETECTED_MODE);
 

From e26bb459d0f3dad83c6a31d5e4480e60760c262b Mon Sep 17 00:00:00 2001
From: Bartosz Juraszewski <bjuraszewski@gmail.com>
Date: Mon, 22 Jun 2026 20:27:33 +0200
Subject: [PATCH 13/19] ASoC: tas2783: Update loaded firmware names to
 linux-firmware 20260519

In linux-firmware commit from 2026-05-19 `2f90f4fe5c67f51a8410907a...`
`ASoC: tas2783: Add Firmware files for tas2783A projects` by Baojun Xu
138 new firmware files for tas2783 were added, none of which are loaded
by the kernel. Kernel expects files to be named with the following
convention: "%04X-%1X-%1X.bin". However the added firmware files follow
"-0x%1X.bin" naming instead with `0x` hex prefix, which fails to load
resulting in following dmesg log:

  slave-tas2783 sdw:0:1:0102:0000:01:8: Direct firmware load for 1714-1-8.bin failed with error -2
  slave-tas2783 sdw:0:1:0102:0000:01:8: Failed to read fw binary 1714-1-8.bin
  slave-tas2783 sdw:0:1:0102:0000:01:b: Direct firmware load for 1714-1-B.bin failed with error -2
  slave-tas2783 sdw:0:1:0102:0000:01:b: Failed to read fw binary 1714-1-B.bin
  slave-tas2783 sdw:0:1:0102:0000:01:8: error playback without fw download
  slave-tas2783 sdw:0:1:0102:0000:01:8: ASoC error (-22): at snd_soc_dai_hw_params() on tas2783-codec

This same commit removes all 22 symlinks from WHENCE, that used naming
without the '0x' prefix to only 6 prevoiusly existing .bin files.

This patch adds `0x` prefix explicitly to the generated firmware name
allowing file to successfully load. In case prefixed firmware is missing
due to out of date linux-firmware, we set the fallback flag and attempt
to load firmware again based on the old file names.

This prefix change results in functioning firmware loading on ASUS
ProArt PX13 HN7306EAC, which uses 1714-1-0x8.bin and 1714-1-0xB.bin
firmware files.

Tested on top of 7.1 and next-20260619 with SND_SOC_AMD_ACP7X set to no.

Signed-off-by: Bartosz Juraszewski <bjuraszewski@gmail.com>
Link: https://patch.msgid.link/20260622182733.23947-1-bjuraszewski@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tas2783-sdw.c | 73 +++++++++++++++++++++++-----------
 1 file changed, 50 insertions(+), 23 deletions(-)

diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c
index 1127ea59b5e4..3d0b116544cc 100644
--- a/sound/soc/codecs/tas2783-sdw.c
+++ b/sound/soc/codecs/tas2783-sdw.c
@@ -100,6 +100,8 @@ struct tas2783_prv {
 	wait_queue_head_t fw_wait;
 	bool fw_dl_task_done;
 	bool fw_dl_success;
+	/* use fallback fw name */
+	bool fw_use_fallback;
 };
 
 static const struct reg_default tas2783_reg_default[] = {
@@ -740,11 +742,19 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context)
 		goto out;
 	}
 
+	/* firmware binary not found*/
 	if (!fmw || !fmw->data) {
-		/* firmware binary not found*/
-		dev_err(tas_dev->dev,
-			"Failed to read fw binary %s\n",
-			tas_dev->rca_binaryname);
+		if (!tas_dev->fw_use_fallback) {
+			tas_dev->fw_use_fallback = true;
+			dev_info(tas_dev->dev,
+				"Failed to read preferred fw binary: %s, attempting fallback binary load\n",
+				tas_dev->rca_binaryname);
+		} else {
+			dev_err(tas_dev->dev,
+				"Failed to read fallback fw binary %s\n",
+				tas_dev->rca_binaryname);
+		}
+
 		ret = -EINVAL;
 		goto out;
 	}
@@ -1105,13 +1115,16 @@ static void tas_generate_fw_name(struct sdw_slave *slave, char *name, size_t siz
 	bool pci_found = false;
 #if IS_ENABLED(CONFIG_PCI)
 	struct device *dev = &slave->dev;
+	struct tas2783_prv *tas_dev = dev_get_drvdata(&slave->dev);
 	struct pci_dev *pci = NULL;
+	const char *fw_uid_prefix = tas_dev->fw_use_fallback ? "" : "0x";
 
 	for (; dev; dev = dev->parent) {
 		if (dev->bus == &pci_bus_type) {
 			pci = to_pci_dev(dev);
-			scnprintf(name, size, "%04X-%1X-%1X.bin",
-				  pci->subsystem_device, bus->link_id, unique_id);
+			scnprintf(name, size, "%04X-%1X-%s%1X.bin",
+				  pci->subsystem_device, bus->link_id,
+				  fw_uid_prefix, unique_id);
 			pci_found = true;
 			break;
 		}
@@ -1123,28 +1136,15 @@ static void tas_generate_fw_name(struct sdw_slave *slave, char *name, size_t siz
 			  bus->link_id, unique_id);
 }
 
-static s32 tas_io_init(struct device *dev, struct sdw_slave *slave)
+static s32 tas_fw_load(struct tas2783_prv *tas_dev, struct sdw_slave *slave)
 {
-	struct tas2783_prv *tas_dev = dev_get_drvdata(dev);
 	s32 ret;
 	u8 unique_id = tas_dev->sdw_peripheral->id.unique_id;
 
-	if (tas_dev->hw_init)
-		return 0;
-
-	tas_dev->fw_dl_task_done = false;
-	tas_dev->fw_dl_success = false;
-
-	ret = regmap_write(tas_dev->regmap, TAS2783_SW_RESET, 0x1);
-	if (ret) {
-		dev_err(dev, "sw reset failed, err=%d", ret);
-		return ret;
-	}
-	usleep_range(2000, 2200);
-
 	tas_generate_fw_name(slave, tas_dev->rca_binaryname,
 			     sizeof(tas_dev->rca_binaryname));
 
+	tas_dev->fw_dl_task_done = false;
 	ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT,
 				      tas_dev->rca_binaryname, tas_dev->dev,
 				      GFP_KERNEL, tas_dev, tas2783_fw_ready);
@@ -1159,8 +1159,35 @@ static s32 tas_io_init(struct device *dev, struct sdw_slave *slave)
 				 msecs_to_jiffies(TIMEOUT_FW_DL_MS));
 	if (!ret) {
 		dev_err(tas_dev->dev, "fw request, wait_event timeout\n");
-		ret = -EAGAIN;
-	} else {
+		return -EAGAIN;
+	}
+
+	return 0;
+}
+
+static s32 tas_io_init(struct device *dev, struct sdw_slave *slave)
+{
+	struct tas2783_prv *tas_dev = dev_get_drvdata(dev);
+	s32 ret;
+
+	if (tas_dev->hw_init)
+		return 0;
+
+	tas_dev->fw_dl_success = false;
+
+	ret = regmap_write(tas_dev->regmap, TAS2783_SW_RESET, 0x1);
+	if (ret) {
+		dev_err(dev, "sw reset failed, err=%d", ret);
+		return ret;
+	}
+	usleep_range(2000, 2200);
+
+	tas_dev->fw_use_fallback = false;
+	ret = tas_fw_load(tas_dev, slave);
+	if (!ret && tas_dev->fw_use_fallback)
+		ret = tas_fw_load(tas_dev, slave);
+
+	if (!ret) {
 		if (tas_dev->sa_func_data)
 			ret = sdca_regmap_write_init(dev, tas_dev->regmap,
 						     tas_dev->sa_func_data);

From d0c415f0076b71b956f62ff6f31de885f0fa2489 Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Tue, 23 Jun 2026 18:25:14 +0800
Subject: [PATCH 14/19] ASoC: rt5575: Use __le32 for SPI burst write address

The addr field in the SPI burst write buffer represents on-wire
little-endian data. Define it as __le32 so that the assignment of
cpu_to_le32() is type correct and avoids sparse endian warnings.

Closes:
https://lore.kernel.org/oe-kbuild-all/202606230139.rFZUVpCa-lkp@intel.com/

Fixes: 420739112e95 ("ASoC: rt5575: Add the codec driver for the ALC5575")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://patch.msgid.link/20260623102514.2422990-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5575-spi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c
index 9a349965435b..d5b3a57c8866 100644
--- a/sound/soc/codecs/rt5575-spi.c
+++ b/sound/soc/codecs/rt5575-spi.c
@@ -17,7 +17,7 @@
 
 struct rt5575_spi_burst_write {
 	u8 cmd;
-	u32 addr;
+	__le32 addr;
 	u8 data[RT5575_SPI_BUF_LEN];
 	u8 dummy;
 } __packed;

From 83d53eca7e5512b53ecef6cf67ea8cbfb595516e Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig=20=28The=20Capable=20Hub=29?=
 <u.kleine-koenig@baylibre.com>
Date: Wed, 24 Jun 2026 10:37:07 +0200
Subject: [PATCH 15/19] ASoC: rockchip: rockchip_sai: #include
 <linux/platform_device.h> explicitly
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Currently that header is only included via:

	<sound/dmaengine_pcm.h> ->
	<sound/soc.h> ->
	<linux/platform_device.h>

which doesn't look reliable, still more in the presence of the comment:

	/* For the current users of sound/soc.h to avoid build issues */

in <sound/soc.h>.

Signed-off-by: Uwe Kleine-König (The Capable Hub) <u.kleine-koenig@baylibre.com>
Reviewed-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Link: https://patch.msgid.link/20260624083708.254517-2-u.kleine-koenig@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_sai.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/rockchip/rockchip_sai.c b/sound/soc/rockchip/rockchip_sai.c
index ed393e5034a4..585e89f61f0d 100644
--- a/sound/soc/rockchip/rockchip_sai.c
+++ b/sound/soc/rockchip/rockchip_sai.c
@@ -11,6 +11,7 @@
 #include <linux/delay.h>
 #include <linux/of_device.h>
 #include <linux/clk.h>
+#include <linux/platform_device.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
 #include <linux/reset.h>

From a7ea04d1ad39d60da397de75b503062ad5fa562b Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Tue, 23 Jun 2026 14:58:21 +0100
Subject: [PATCH 16/19] ASoC: soc-core: Don't fail if device_link could not be
 created

If device_link_add() fails in snd_soc_bind_card() just skip that
driver pair and carry on.

This means that ASoC must now keep track of which components it was
able to device_link to the card->dev. The new card_device_link member
of struct snd_soc_component is non-NULL if a device_link exists.

The intent of the device link is to ensure that the machine driver
system-suspends before the component drivers, to prevent ASoC
suspend attempting to reconfigure a driver that has already suspended.

It isn't possible to create this device link if the machine driver is
a parent of the component driver or already has a device_link in the
opposite direction. In this case skip the link. A warn is placed in
kernel log since this might indicate a genuine design problem with
those two drivers (this can be downgraded to dbg in future when
people are happy that all these special drivers correctly handle their
reversed shutdown order).

Fixes: 0f54ce994b23 ("ASoC: soc-core: Create device_link to ensure correct suspend order")
Reported-by: Marek Szyprowski <m.szyprowski@samsung.com>
Closes: https://lore.kernel.org/linux-sound/61bd38e7-5eb9-4448-a93f-afa2ccbd1c9d@opensource.cirrus.com/T/#m496fe5a11b0a3649afd2e85da5e1cea82bb16d8a
Tested-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20260623135821.4125543-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-component.h |  4 ++++
 sound/soc/soc-core.c          | 35 +++++++++++++++--------------------
 2 files changed, 19 insertions(+), 20 deletions(-)

diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 11bc9527653f..aa423865dbe7 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -10,6 +10,8 @@
 
 #include <sound/soc.h>
 
+struct device_link;
+
 /*
  * Component probe and remove ordering levels for components with runtime
  * dependencies.
@@ -216,6 +218,8 @@ struct snd_soc_component {
 	struct list_head card_aux_list; /* for auxiliary bound components */
 	struct list_head card_list;
 
+	struct device_link *card_device_link;
+
 	const struct snd_soc_component_driver *driver;
 
 	struct list_head dai_list;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 86b6c752a56b..7817beea5b3b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1980,19 +1980,15 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card)
 	}
 }
 
-static void snd_soc_remove_device_links(struct snd_soc_card *card,
-					struct snd_soc_component *stop_at)
+static void snd_soc_remove_device_links(struct snd_soc_card *card)
 {
 	struct snd_soc_component *component;
 
 	for_each_card_components(card, component) {
-		if (card->dev == component->dev)
-			continue;
-
-		device_link_remove(card->dev, component->dev);
-
-		if (component == stop_at)
-			return;
+		if (component->card_device_link) {
+			device_link_del(component->card_device_link);
+			component->card_device_link = NULL;
+		}
 	}
 }
 
@@ -2001,7 +1997,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card)
 	if (snd_soc_card_is_instantiated(card)) {
 		card->instantiated = false;
 
-		snd_soc_remove_device_links(card, NULL);
+		snd_soc_remove_device_links(card);
 
 		soc_cleanup_card_resources(card);
 	}
@@ -2012,7 +2008,6 @@ static int snd_soc_bind_card(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_component *component;
 	struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
-	struct snd_soc_component *last_devlinked_component = NULL;
 	int ret;
 
 	snd_soc_card_mutex_lock_root(card);
@@ -2141,19 +2136,21 @@ static int snd_soc_bind_card(struct snd_soc_card *card)
 	 * Add device_link from card to component so that system_suspend
 	 * will be done in the correct order. The card must suspend first
 	 * to stop audio activity before the components suspend.
+	 *
+	 * If a driver pair already have a link in the opposite direction
+	 * they must manage their own suspend order.
 	 */
 	for_each_card_components(card, component) {
 		if (card->dev == component->dev)
 			continue;
 
-		if (!device_link_add(card->dev, component->dev, DL_FLAG_STATELESS)) {
-			dev_warn(card->dev, "Failed to create device link to %s\n",
+		component->card_device_link = device_link_add(card->dev,
+							      component->dev,
+							      DL_FLAG_STATELESS);
+		if (!component->card_device_link) {
+			dev_warn(card->dev, "Could not create device link to %s\n",
 				 dev_name(component->dev));
-			ret = -EINVAL;
-			goto probe_end;
 		}
-
-		last_devlinked_component = component;
 	}
 
 	ret = snd_soc_card_late_probe(card);
@@ -2185,9 +2182,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card)
 
 probe_end:
 	if (ret < 0) {
-		if (last_devlinked_component)
-			snd_soc_remove_device_links(card, last_devlinked_component);
-
+		snd_soc_remove_device_links(card);
 		soc_cleanup_card_resources(card);
 	}
 

From 64184f07e7516eb3ac1839387eb7cbe2dbeaa96f Mon Sep 17 00:00:00 2001
From: Baojun Xu <baojun.xu@ti.com>
Date: Thu, 25 Jun 2026 18:28:15 +0800
Subject: [PATCH 17/19] ASoC: tas2781: Update default register address to
 TAS2563

The previous TAS2781 address is overwritten during calibration,
so switching to the TAS2563 address is required.

Signed-off-by: Baojun Xu <baojun.xu@ti.com>
Link: https://patch.msgid.link/20260625102815.20043-1-baojun.xu@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/tas2781.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h
index d45f805b5d18..b763da613769 100644
--- a/include/sound/tas2781.h
+++ b/include/sound/tas2781.h
@@ -59,9 +59,9 @@
 #define TASDEVICE_CHECKSUM_REG		TASDEVICE_REG(0x0, 0x0, 0x7e)
 
 /* XM_340 */
-#define	TASDEVICE_XM_A1_REG	TASDEVICE_REG(0x64, 0x63, 0x3c)
+#define	TASDEVICE_XM_A1_REG	TASDEVICE_REG(0x64, 0x02, 0x4c)
 /* XM_341 */
-#define	TASDEVICE_XM_A2_REG	TASDEVICE_REG(0x64, 0x63, 0x38)
+#define	TASDEVICE_XM_A2_REG	TASDEVICE_REG(0x64, 0x02, 0x64)
 
 /* Volume control */
 #define TAS2563_DVC_LVL			TASDEVICE_REG(0x00, 0x02, 0x0c)

From c5315c8dcbcb9271f59662104428cd9229c9a7cb Mon Sep 17 00:00:00 2001
From: Christian Hewitt <christianshewitt@gmail.com>
Date: Thu, 25 Jun 2026 12:28:11 +0000
Subject: [PATCH 18/19] ASoC: codecs: pcm512x: only print info once on no sclk

If sclk is not provided the driver falls back to using bclk and prints
an info message in the system log. Under normal operations the message
is repeated many times:

[   17.929576] pcm512x 0-004c: No SCLK, using BCLK: -2
[   17.949172] pcm512x 0-004c: No SCLK, using BCLK: -2
[   17.953029] pcm512x 0-004c: No SCLK, using BCLK: -2
[   17.965059] pcm512x 0-004c: No SCLK, using BCLK: -2
[   82.592980] pcm512x 0-004c: No SCLK, using BCLK: -2
[   82.866293] pcm512x 0-004c: No SCLK, using BCLK: -2

Switch from dev_info to dev_info_once to reduce log noise.

Signed-off-by: Christian Hewitt <christianshewitt@gmail.com>
Link: https://patch.msgid.link/20260625122811.4056274-1-christianshewitt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm512x.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index fdef98ce52f1..fe3b5011fa16 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -633,7 +633,7 @@ static int pcm512x_dai_startup_slave(struct snd_pcm_substream *substream,
 	struct regmap *regmap = pcm512x->regmap;
 
 	if (IS_ERR(pcm512x->sclk)) {
-		dev_info(dev, "No SCLK, using BCLK: %ld\n",
+		dev_info_once(dev, "No SCLK, using BCLK: %ld\n",
 			 PTR_ERR(pcm512x->sclk));
 
 		/* Disable reporting of missing SCLK as an error */

From cf6f56990ea21172e085f0588e5bbf2089ce8f58 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Thu, 25 Jun 2026 18:24:16 +0800
Subject: [PATCH 19/19] ASoC: fsl_asrc_dma: fix eDMA maxburst misalignment with
 channel count

The back-end consumes data in units of the number of channels. When the
maxburst value is not evenly divisible by the channel count, the DMA
transfer length does not align with the FIFO frame boundary, causing
wrong data to be copied and audible noise at the end of the stream.

This is specific to eDMA: eDMA only responds to DMA requests from the
back-end, whereas SDMA handles requests from both the front-end and the
back-end and is not affected.

For eDMA, when the back-end maxburst is not evenly divisible by the
channel count, align it to the nearest valid boundary:
- If maxburst >= channel count, override to the channel count so each
  transfer corresponds to exactly one audio frame.
- If maxburst < channel count, override to 1 to avoid partial-frame
  transfers.

Retain the original maxburst for SDMA or when it already aligns with
the channel count.

Fixes: c05f10f28ef6 ("ASoC: fsl_asrc: Add support for imx8qm & imx8qxp")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20260625102416.424911-1-shengjiu.wang@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc_dma.c | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)

diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 5aa96af994c4..38f2b7c63133 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -288,6 +288,26 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	config_be.dst_addr_width = buswidth;
 	config_be.dst_maxburst = dma_params_be->maxburst;
 
+	/*
+	 * For eDMA, the back-end may report a maxburst size that is not evenly
+	 * divisible by the channel count. This causes the DMA transfer length
+	 * to misalign with the FIFO boundary, resulting in wrong data and
+	 * audible noise. Align maxburst to the nearest valid boundary:
+	 * - If maxburst >= channel count, override to the channel count so
+	 *   each transfer equals exactly one audio frame.
+	 * - If maxburst < channel count, override to 1 to avoid partial-frame
+	 *   transfers.
+	 */
+	if (asrc->use_edma && (dma_params_be->maxburst % params_channels(params))) {
+		if (dma_params_be->maxburst >= params_channels(params)) {
+			config_be.src_maxburst = params_channels(params);
+			config_be.dst_maxburst = params_channels(params);
+		} else {
+			config_be.src_maxburst = 1;
+			config_be.dst_maxburst = 1;
+		}
+	}
+
 	memset(&audio_config, 0, sizeof(audio_config));
 	config_be.peripheral_config = &audio_config;
 	config_be.peripheral_size  = sizeof(audio_config);