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authorTakashi Iwai <tiwai@suse.de>2024-07-25 18:04:55 +0200
committerTakashi Iwai <tiwai@suse.de>2024-07-25 18:04:55 +0200
commite8b96a66ae01d039699bac256c5b6b30b2284170 (patch)
treef4a7efe2406bc0f6ae37cedb4241765f0669f037
parentALSA: hda/realtek: Implement sound init sequence for Samsung Galaxy Book3 Pro... (diff)
parentASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components (diff)
downloadlinux-e8b96a66ae01d039699bac256c5b6b30b2284170.tar.xz
linux-e8b96a66ae01d039699bac256c5b6b30b2284170.zip
Merge tag 'asoc-fix-v6.11-merge-window' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.11 A selection of routine fixes and quirks that came in since the merge window. The fsl-asoc-card change is a fix for systems with multiple cards where updating templates in place leaks data from one card to another.
-rw-r--r--sound/soc/amd/yc/acp6x-mach.c7
-rw-r--r--sound/soc/codecs/tas2781-fmwlib.c2
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c46
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-ssp-common.c9
-rw-r--r--sound/soc/intel/common/soc-intel-quirks.h2
-rw-r--r--sound/soc/sof/amd/pci-vangogh.c1
-rw-r--r--sound/soc/sof/imx/imx8m.c2
-rw-r--r--sound/soc/sof/intel/hda-loader.c20
-rw-r--r--sound/soc/sof/intel/hda.c17
-rw-r--r--sound/soc/sof/ipc4-topology.c18
-rw-r--r--sound/soc/tegra/Kconfig1
11 files changed, 85 insertions, 40 deletions
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index f54466ed8e3e..1769e07e83dc 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -224,6 +224,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
.driver_data = &acp6x_card,
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "21M5"),
+ }
+ },
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_NAME, "82QF"),
}
},
diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c
index 63626b982d04..8f9a3ae7153e 100644
--- a/sound/soc/codecs/tas2781-fmwlib.c
+++ b/sound/soc/codecs/tas2781-fmwlib.c
@@ -2162,7 +2162,7 @@ static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i)
return;
cal = cal_fmw->calibrations;
- if (cal)
+ if (!cal)
return;
load_calib_data(priv, &cal->dev_data);
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 82df887b3af5..f6c3aeff0d8e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -306,27 +306,12 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
-SND_SOC_DAILINK_DEFS(hifi,
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(hifi_fe,
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_DUMMY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(hifi_be,
- DAILINK_COMP_ARRAY(COMP_EMPTY()),
- DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()));
-
static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
/* Default ASoC DAI Link*/
{
.name = "HiFi",
.stream_name = "HiFi",
.ops = &fsl_asoc_card_ops,
- SND_SOC_DAILINK_REG(hifi),
},
/* DPCM Link between Front-End and Back-End (Optional) */
{
@@ -335,7 +320,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
.dynamic = 1,
- SND_SOC_DAILINK_REG(hifi_fe),
},
{
.name = "HiFi-ASRC-BE",
@@ -345,7 +329,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
.no_pcm = 1,
- SND_SOC_DAILINK_REG(hifi_be),
},
};
@@ -637,6 +620,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
struct device *codec_dev[2] = { NULL, NULL };
+ struct snd_soc_dai_link_component *dlc;
const char *codec_dai_name[2];
const char *codec_dev_name[2];
u32 asrc_fmt = 0;
@@ -717,7 +701,35 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
memcpy(priv->dai_link, fsl_asoc_card_dai,
sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+ /*
+ * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms
+ * "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms
+ * "DPCM Link Back-End": 1 cpus, 2 codecs
+ * totally 10 components
+ */
+ dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL);
+ if (!dlc) {
+ ret = -ENOMEM;
+ goto asrc_fail;
+ }
+
+ priv->dai_link[0].cpus = &dlc[0];
+ priv->dai_link[0].num_cpus = 1;
+ priv->dai_link[0].codecs = &dlc[1];
priv->dai_link[0].num_codecs = 1;
+ priv->dai_link[0].platforms = &dlc[3];
+ priv->dai_link[0].num_platforms = 1;
+
+ priv->dai_link[1].cpus = &dlc[4];
+ priv->dai_link[1].num_cpus = 1;
+ priv->dai_link[1].codecs = &dlc[5];
+ priv->dai_link[1].num_codecs = 0; /* dummy */
+ priv->dai_link[1].platforms = &dlc[6];
+ priv->dai_link[1].num_platforms = 1;
+
+ priv->dai_link[2].cpus = &dlc[7];
+ priv->dai_link[2].num_cpus = 1;
+ priv->dai_link[2].codecs = &dlc[8];
priv->dai_link[2].num_codecs = 1;
priv->card.dapm_routes = audio_map;
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 75d0b931d895..de7a3f7f47f1 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -64,6 +64,15 @@ static const struct codec_map amps[] = {
CODEC_MAP_ENTRY("RT1015P", "rt1015", RT1015P_ACPI_HID, CODEC_RT1015P),
CODEC_MAP_ENTRY("RT1019P", "rt1019", RT1019P_ACPI_HID, CODEC_RT1019P),
CODEC_MAP_ENTRY("RT1308", "rt1308", RT1308_ACPI_HID, CODEC_RT1308),
+
+ /*
+ * Monolithic components
+ *
+ * Only put components that can serve as both the amp and the codec below this line.
+ * This will ensure that if the part is used just as a codec and there is an amp as well
+ * then the amp will be selected properly.
+ */
+ CODEC_MAP_ENTRY("RT5650", "rt5650", RT5650_ACPI_HID, CODEC_RT5650),
};
enum snd_soc_acpi_intel_codec
diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h
index de4e550c5b34..42bd51456b94 100644
--- a/sound/soc/intel/common/soc-intel-quirks.h
+++ b/sound/soc/intel/common/soc-intel-quirks.h
@@ -11,7 +11,7 @@
#include <linux/platform_data/x86/soc.h>
-#if IS_ENABLED(CONFIG_X86)
+#if IS_REACHABLE(CONFIG_IOSF_MBI)
#include <linux/dmi.h>
#include <asm/iosf_mbi.h>
diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c
index 16eb2994fbab..eba580840100 100644
--- a/sound/soc/sof/amd/pci-vangogh.c
+++ b/sound/soc/sof/amd/pci-vangogh.c
@@ -34,7 +34,6 @@ static const struct sof_amd_acp_desc vangogh_chip_info = {
.dsp_intr_base = ACP5X_DSP_SW_INTR_BASE,
.sram_pte_offset = ACP5X_SRAM_PTE_OFFSET,
.hw_semaphore_offset = ACP5X_AXI2DAGB_SEM_0,
- .acp_clkmux_sel = ACP5X_CLKMUX_SEL,
.probe_reg_offset = ACP5X_FUTURE_REG_ACLK_0,
};
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index 1c7019c3cbd3..cdd1e79ef9f6 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -234,7 +234,7 @@ static int imx8m_probe(struct snd_sof_dev *sdev)
/* set default mailbox offset for FW ready message */
sdev->dsp_box.offset = MBOX_OFFSET;
- priv->regmap = syscon_regmap_lookup_by_compatible("fsl,dsp-ctrl");
+ priv->regmap = syscon_regmap_lookup_by_phandle(np, "fsl,dsp-ctrl");
if (IS_ERR(priv->regmap)) {
dev_err(sdev->dev, "cannot find dsp-ctrl registers");
ret = PTR_ERR(priv->regmap);
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index b8b914eaf7e0..75f6240cf3e1 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -310,15 +310,19 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
return ret;
}
- /* Wait for completion of transfer */
- time_left = wait_for_completion_timeout(&hda_stream->ioc,
- msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
-
- if (!time_left) {
- dev_err(sdev->dev, "Code loader DMA did not complete\n");
- return -ETIMEDOUT;
+ if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) {
+ /* Wait for completion of transfer */
+ time_left = wait_for_completion_timeout(&hda_stream->ioc,
+ msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
+
+ if (!time_left) {
+ dev_err(sdev->dev, "Code loader DMA did not complete\n");
+ return -ETIMEDOUT;
+ }
+ dev_dbg(sdev->dev, "Code loader DMA done\n");
}
- dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n");
+
+ dev_dbg(sdev->dev, "waiting for FW_ENTERED status\n");
status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
chip->rom_status_reg, reg,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index daf364f773dd..5a40b8fbbbd3 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1307,9 +1307,10 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
const struct sof_dev_desc *desc = sof_pdata->desc;
struct hdac_bus *bus = sof_to_bus(sdev);
struct snd_soc_acpi_mach *mach = NULL;
- enum snd_soc_acpi_intel_codec codec_type;
+ enum snd_soc_acpi_intel_codec codec_type, amp_type;
const char *tplg_filename;
const char *tplg_suffix;
+ bool amp_name_valid;
/* Try I2S or DMIC if it is supported */
if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC)))
@@ -1413,15 +1414,16 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
}
}
- codec_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev);
+ amp_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev);
+ codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev);
+ amp_name_valid = amp_type != CODEC_NONE && amp_type != codec_type;
- if (tplg_fixup &&
- mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME &&
- codec_type != CODEC_NONE) {
- tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(codec_type);
+ if (tplg_fixup && amp_name_valid &&
+ mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME) {
+ tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(amp_type);
if (!tplg_suffix) {
dev_err(sdev->dev, "no tplg suffix found, amp %d\n",
- codec_type);
+ amp_type);
return NULL;
}
@@ -1436,7 +1438,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
add_extension = true;
}
- codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev);
if (tplg_fixup &&
mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME &&
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 90f6856ee80c..87be7f16e8c2 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1358,7 +1358,13 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
ipc4_copier = dai->private;
if (pipeline->use_chain_dma) {
- pipeline->msg.primary = 0;
+ /*
+ * Preserve the DMA Link ID and clear other bits since
+ * the DMA Link ID is only configured once during
+ * dai_config, other fields are expected to be 0 for
+ * re-configuration
+ */
+ pipeline->msg.primary &= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
pipeline->msg.extension = 0;
}
@@ -3095,8 +3101,14 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
return 0;
if (pipeline->use_chain_dma) {
- pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
- pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data);
+ /*
+ * Only configure the DMA Link ID for ChainDMA when this op is
+ * invoked with SOF_DAI_CONFIG_FLAGS_HW_PARAMS
+ */
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+ pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
+ pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data);
+ }
return 0;
}
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 74effc57a7a0..2463c22e9cf6 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -78,6 +78,7 @@ config SND_SOC_TEGRA210_DMIC
config SND_SOC_TEGRA210_I2S
tristate "Tegra210 I2S module"
+ select SND_SIMPLE_CARD_UTILS
help
Config to enable the Inter-IC Sound (I2S) Controller which
implements full-duplex and bidirectional and single direction