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Date:   Fri, 21 Apr 2023 10:43:46 +0200
Message-ID: <87pm7xsjzx.wl-tiwai@suse.de>
From:   Takashi Iwai <tiwai@suse.de>
To:     Linus Torvalds <torvalds@linux-foundation.org>
Cc:     Mark Brown <broonie@kernel.org>,
        Liam Girdwood <lgirdwood@gmail.com>,
        Linux Kernel Mailing List <linux-kernel@vger.kernel.org>
Subject: [GIT PULL] sound fixes for 6.3
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Linus,

please pull sound fixes for v6.3 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-6.3

The topmost commit is 2ae147d643d326f74d93ba4f72a405f25f2677ea

----------------------------------------------------------------

sound fixes for 6.3-final

Just a few fixes: all small and device-specific (ASoC FSL, SOF, and
HD-audio quirks), should be safe to apply at the last minute.

----------------------------------------------------------------

Andy Chi (1):
      ALSA: hda/realtek: fix mute/micmute LEDs for a HP ProBook

Chancel Liu (1):
      ASoC: fsl_sai: Fix pins setting for i.MX8QM platform

Daniel Baluta (1):
      ASoC: SOF: pm: Tear down pipelines only if DSP was active

Jaroslav Kysela (1):
      ALSA: hda/realtek: Remove specific patch for Dell Precision 3260

Long Wang (1):
      ASoC: max98373: change power down sequence for smart amp

Nikita Zhandarovich (1):
      ASoC: fsl_asrc_dma: fix potential null-ptr-deref

Peter Ujfalusi (1):
      ASoC: SOF: ipc4-topology: Clarify bind failure caused by missing fw_module

---
 sound/pci/hda/patch_realtek.c |  2 +-
 sound/soc/codecs/max98373.c   |  4 ++--
 sound/soc/fsl/fsl_asrc_dma.c  | 11 ++++++++---
 sound/soc/fsl/fsl_sai.c       |  2 +-
 sound/soc/sof/ipc4-topology.c | 10 ++++++----
 sound/soc/sof/pm.c            |  8 +++++++-
 6 files changed, 25 insertions(+), 12 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3b9f077a227f..f70d6a33421d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9288,7 +9288,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
@@ -9469,6 +9468,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED),
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index f90a6a7ba83b..fde055c6c894 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -31,7 +31,7 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w,
 			MAX98373_GLOBAL_EN_MASK, 1);
 		usleep_range(30000, 31000);
 		break;
-	case SND_SOC_DAPM_POST_PMD:
+	case SND_SOC_DAPM_PRE_PMD:
 		regmap_update_bits(max98373->regmap,
 			MAX98373_R20FF_GLOBAL_SHDN,
 			MAX98373_GLOBAL_EN_MASK, 0);
@@ -64,7 +64,7 @@ static const struct snd_kcontrol_new max98373_spkfb_control =
 static const struct snd_soc_dapm_widget max98373_dapm_widgets[] = {
 SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback",
 	MAX98373_R202B_PCM_RX_EN, 0, 0, max98373_dac_event,
-	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0,
 	&max98373_dai_controls),
 SND_SOC_DAPM_OUTPUT("BE_OUT"),
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 3b81a465814a..05a7d1588d20 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -209,14 +209,19 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 		be_chan = soc_component_to_pcm(component_be)->chan[substream->stream];
 		tmp_chan = be_chan;
 	}
-	if (!tmp_chan)
-		tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
+	if (!tmp_chan) {
+		tmp_chan = dma_request_chan(dev_be, tx ? "tx" : "rx");
+		if (IS_ERR(tmp_chan)) {
+			dev_err(dev, "failed to request DMA channel for Back-End\n");
+			return -EINVAL;
+		}
+	}
 
 	/*
 	 * An EDMA DEV_TO_DEV channel is fixed and bound with DMA event of each
 	 * peripheral, unlike SDMA channel that is allocated dynamically. So no
 	 * need to configure dma_request and dma_request2, but get dma_chan of
-	 * Back-End device directly via dma_request_slave_channel.
+	 * Back-End device directly via dma_request_chan.
 	 */
 	if (!asrc->use_edma) {
 		/* Get DMA request of Back-End */
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1b197478b3d9..990bba0be1fb 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1546,7 +1546,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
 	.use_imx_pcm = true,
 	.use_edma = true,
 	.fifo_depth = 64,
-	.pins = 1,
+	.pins = 4,
 	.reg_offset = 0,
 	.mclk0_is_mclk1 = false,
 	.flags = 0,
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 669b99a4f76e..3a5394c3dd83 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1806,10 +1806,12 @@ static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route *
 	int ret;
 
 	if (!src_fw_module || !sink_fw_module) {
-		/* The NULL module will print as "(efault)" */
-		dev_err(sdev->dev, "source %s or sink %s widget weren't set up properly\n",
-			src_fw_module->man4_module_entry.name,
-			sink_fw_module->man4_module_entry.name);
+		dev_err(sdev->dev,
+			"cannot bind %s -> %s, no firmware module for: %s%s\n",
+			src_widget->widget->name, sink_widget->widget->name,
+			src_fw_module ? "" : " source",
+			sink_fw_module ? "" : " sink");
+
 		return -ENODEV;
 	}
 
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 8d3383085d12..85412aeb1ca1 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -183,6 +183,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg);
 	pm_message_t pm_state;
 	u32 target_state = snd_sof_dsp_power_target(sdev);
+	u32 old_state = sdev->dsp_power_state.state;
 	int ret;
 
 	/* do nothing if dsp suspend callback is not set */
@@ -192,7 +193,12 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
 		return 0;
 
-	if (tplg_ops && tplg_ops->tear_down_all_pipelines)
+	/* we need to tear down pipelines only if the DSP hardware is
+	 * active, which happens for PCI devices. if the device is
+	 * suspended, it is brought back to full power and then
+	 * suspended again
+	 */
+	if (tplg_ops && tplg_ops->tear_down_all_pipelines && (old_state == SOF_DSP_PM_D0))
 		tplg_ops->tear_down_all_pipelines(sdev, false);
 
 	if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)