Merge tag 'v5.4.50' into 5.4-2.1.x-imx

This is the 5.4.50 stable release

Conflicts (manual resolve):

arch/arm64/boot/dts/freescale/imx8mm-evk.dts:
arch/arm64/boot/dts/freescale/imx8mn-ddr4-evk.dts:
NXP patches kept to provide proper LDO setup:
imx8mm-evk.dts: 975d8ab07267ded741c4c5d7500e524c85ab40d3
imx8mn-ddr4-evk.dts: e8e35fd0e759965809f3dca5979a908a09286198

drivers/net/ethernet/freescale/enetc/enetc.c
Commit [ce06fcb6a66d22b080385768113ee28bf3593984] from upstream merged,
base NXP version kept

drivers/usb/cdns3/ep0.c
Manual merge of commit [be8df027079b856a89e85778330394cea2b0b290] from
upstream to protect cdns3_check_new_setup

Signed-off-by: Andrey Zhizhikin <andrey.z@gmail.com>

6 files changed, 37 insertions, 10 deletions

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index ee3df508dbfe..a7d50d60cadb 100755
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -681,8 +681,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	struct regmap *regs = ssi->regs;
 	u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
 	unsigned long clkrate, baudrate, tmprate;
-	unsigned int slots = params_channels(hw_params);
-	unsigned int slot_width = 32;
+	unsigned int channels = params_channels(hw_params);
+	unsigned int slot_width = params_width(hw_params);
+	unsigned int slots = 2;
 	u64 sub, savesub = 100000;
 	unsigned int freq;
 	bool baudclk_is_used;
@@ -691,10 +692,14 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	/* Override slots and slot_width if being specifically set... */
 	if (ssi->slots)
 		slots = ssi->slots;
-	/* ...but keep 32 bits if slots is 2 -- I2S Master mode */
-	if (ssi->slot_width && slots != 2)
+	if (ssi->slot_width)
 		slot_width = ssi->slot_width;
 
+	/* ...but force 32 bits for stereo audio using I2S Master Mode */
+	if (channels == 2 &&
+	    (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER)
+		slot_width = 32;
+
 	/* Generate bit clock based on the slot number and slot width */
 	freq = slots * slot_width * params_rate(hw_params);
 
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 6c20bdd850f3..8ada4ecba847 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -4,6 +4,7 @@
 
 #include <linux/module.h>
 #include "common.h"
+#include "qdsp6/q6afe.h"
 
 int qcom_snd_parse_of(struct snd_soc_card *card)
 {
@@ -101,6 +102,15 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			}
 			link->no_pcm = 1;
 			link->ignore_pmdown_time = 1;
+
+			if (q6afe_is_rx_port(link->id)) {
+				link->dpcm_playback = 1;
+				link->dpcm_capture = 0;
+			} else {
+				link->dpcm_playback = 0;
+				link->dpcm_capture = 1;
+			}
+
 		} else {
 			dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL);
 			if (!dlc)
@@ -113,12 +123,12 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			link->codecs->dai_name = "snd-soc-dummy-dai";
 			link->codecs->name = "snd-soc-dummy";
 			link->dynamic = 1;
+			link->dpcm_playback = 1;
+			link->dpcm_capture = 1;
 		}
 
 		link->ignore_suspend = 1;
 		link->nonatomic = 1;
-		link->dpcm_playback = 1;
-		link->dpcm_capture = 1;
 		link->stream_name = link->name;
 		link++;
 
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7a58c8..0ce4eb60f984 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -800,6 +800,14 @@ int q6afe_get_port_id(int index)
 }
 EXPORT_SYMBOL_GPL(q6afe_get_port_id);
 
+int q6afe_is_rx_port(int index)
+{
+	if (index < 0 || index >= AFE_PORT_MAX)
+		return -EINVAL;
+
+	return port_maps[index].is_rx;
+}
+EXPORT_SYMBOL_GPL(q6afe_is_rx_port);
 static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
 			    struct q6afe_port *port)
 {
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed5422baff..1a0f80a14afe 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port);
 int q6afe_port_stop(struct q6afe_port *port);
 void q6afe_port_put(struct q6afe_port *port);
 int q6afe_get_port_id(int index);
+int q6afe_is_rx_port(int index);
 void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 			    struct q6afe_hdmi_cfg *cfg);
 void q6afe_slim_port_prepare(struct q6afe_port *port,
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a33a55e..835ac98a789c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
 #define ASM_STREAM_CMD_FLUSH			0x00010BCE
 #define ASM_SESSION_CMD_PAUSE			0x00010BD3
 #define ASM_DATA_CMD_EOS			0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS		0x00010C1C
 #define ASM_NULL_POPP_TOPOLOGY			0x00010C68
 #define ASM_STREAM_CMD_FLUSH_READBUFS		0x00010C09
 #define ASM_STREAM_CMD_SET_ENCDEC_PARAM		0x00010C10
@@ -546,9 +547,6 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		case ASM_SESSION_CMD_SUSPEND:
 			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
 			break;
-		case ASM_DATA_CMD_EOS:
-			client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
-			break;
 		case ASM_STREAM_CMD_FLUSH:
 			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
 			break;
@@ -652,6 +650,9 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		}
 
 		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+		break;
 	}
 
 	if (ac->cb)
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 7cd42fcfcf38..1707414cfa92 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -590,8 +590,10 @@ static int rockchip_pdm_resume(struct device *dev)
 	int ret;
 
 	ret = pm_runtime_get_sync(dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put(dev);
 		return ret;
+	}
 
 	ret = regcache_sync(pdm->regmap);