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authorLinus Torvalds <torvalds@linux-foundation.org>2023-09-08 13:07:50 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2023-09-08 13:07:50 -0700
commita3d231e44ab827345d24973a621e9364949423a4 (patch)
treebc1b5c45b7441d0b28bdbf7707088af13cc31faa
parentca9c7abf9502e108fae0e34181e01b1a20bc439f (diff)
parentecc8b4d0b6e19031733f28537971f8c5088dada2 (diff)
Merge tag 'sound-fix-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "A collection of fixes for 6.6-rc1. All small and easy ones. - The corrections of the previous PCM iov_iter transitions - Regression fixes in MIDI 2.0 / USB changes - Various ASoC codec fixes for Cirrus, Realtek, WCD - ASoC AMD quirks and ASoC Intel AVS driver workaround" * tag 'sound-fix-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits) ALSA: hda/realtek - ALC287 I2S speaker platform support ASoC: amd: yc: Fix a non-functional mic on Lenovo 82TL ASoC: Intel: avs: Provide support for fallback topology ALSA: seq: Fix snd_seq_expand_var_event() call to user-space ALSA: usb-audio: Fix potential memory leaks at error path for UMP open ALSA: hda/cirrus: Fix broken audio on hardware with two CS42L42 codecs. ASoC: rt5645: NULL pointer access when removing jack ASoC: amd: yc: Add DMI entries to support Victus by HP Gaming Laptop 15-fb0xxx (8A3E) MAINTAINERS: Update the MAINTAINERS enties for TEXAS INSTRUMENTS ASoC DRIVERS ALSA: sb: Fix wrong argument in commented code ALSA: pcm: Fix error checks of default read/write copy ops ASoC: Name iov_iter argument as iterator instead of buffer ASoC: dmaengine: Drop unused iov_iter for process callback ALSA: hda/tas2781: Use standard clamp() macro ASoC: cs35l56: Waiting for firmware to boot must be tolerant of I/O errors ASoC: dt-bindings: fsl_easrc: Add support for imx8mp-easrc ASoC: cs42l43: Fix missing error code in cs42l43_codec_probe() ASoC: cs35l45: Rename DACPCM1 Source control ASoC: cs35l45: Fix "Dead assigment" warning ASoC: cs35l45: Add support for Chip ID 0x35A460 ...
-rw-r--r--Documentation/devicetree/bindings/sound/fsl,easrc.yaml8
-rw-r--r--MAINTAINERS2
-rw-r--r--include/sound/dmaengine_pcm.h2
-rw-r--r--include/sound/soc-component.h4
-rw-r--r--sound/core/pcm_lib.c8
-rw-r--r--sound/core/seq/seq_memory.c9
-rw-r--r--sound/isa/sb/emu8000_pcm.c2
-rw-r--r--sound/pci/hda/patch_cs8409.c2
-rw-r--r--sound/pci/hda/patch_cs8409.h1
-rw-r--r--sound/pci/hda/patch_realtek.c30
-rw-r--r--sound/pci/hda/tas2781_hda_i2c.c16
-rw-r--r--sound/soc/amd/yc/acp6x-mach.c14
-rw-r--r--sound/soc/atmel/mchp-pdmc.c2
-rw-r--r--sound/soc/codecs/Kconfig6
-rw-r--r--sound/soc/codecs/Makefile8
-rw-r--r--sound/soc/codecs/cs35l45.c11
-rw-r--r--sound/soc/codecs/cs35l56-shared.c29
-rw-r--r--sound/soc/codecs/cs42l43.c3
-rw-r--r--sound/soc/codecs/rt5645.c16
-rw-r--r--sound/soc/codecs/wcd-clsh-v2.c8
-rw-r--r--sound/soc/intel/avs/pcm.c22
-rw-r--r--sound/soc/soc-component.c4
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c10
-rw-r--r--sound/soc/stm/stm32_sai_sub.c2
-rw-r--r--sound/usb/midi2.c7
25 files changed, 160 insertions, 66 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml
index bdde68a1059c..a680d7aff237 100644
--- a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml
@@ -14,7 +14,13 @@ properties:
pattern: "^easrc@.*"
compatible:
- const: fsl,imx8mn-easrc
+ oneOf:
+ - enum:
+ - fsl,imx8mn-easrc
+ - items:
+ - enum:
+ - fsl,imx8mp-easrc
+ - const: fsl,imx8mn-easrc
reg:
maxItems: 1
diff --git a/MAINTAINERS b/MAINTAINERS
index 389fe9e38884..a62f5a2a1c9e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -21244,7 +21244,7 @@ F: sound/soc/ti/
TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
M: Shenghao Ding <shenghao-ding@ti.com>
M: Kevin Lu <kevin-lu@ti.com>
-M: Baojun Xu <x1077012@ti.com>
+M: Baojun Xu <baojun.xu@ti.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
F: Documentation/devicetree/bindings/sound/tas2552.txt
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index c9a8bce9a785..d70c55f17df7 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -142,7 +142,7 @@ struct snd_dmaengine_pcm_config {
struct snd_pcm_substream *substream);
int (*process)(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- struct iov_iter *buf, unsigned long bytes);
+ unsigned long bytes);
dma_filter_fn compat_filter_fn;
struct device *dma_dev;
const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 17bea3144551..ceca69b46a82 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -139,7 +139,7 @@ struct snd_soc_component_driver {
struct snd_pcm_audio_tstamp_report *audio_tstamp_report);
int (*copy)(struct snd_soc_component *component,
struct snd_pcm_substream *substream, int channel,
- unsigned long pos, struct iov_iter *buf,
+ unsigned long pos, struct iov_iter *iter,
unsigned long bytes);
struct page *(*page)(struct snd_soc_component *component,
struct snd_pcm_substream *substream,
@@ -511,7 +511,7 @@ int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream);
int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
int channel, unsigned long pos,
- struct iov_iter *buf, unsigned long bytes);
+ struct iov_iter *iter, unsigned long bytes);
struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
unsigned long offset);
int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4859fb1caec9..a11cd7d6295f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1992,8 +1992,8 @@ static int default_write_copy(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
struct iov_iter *iter, unsigned long bytes)
{
- if (!copy_from_iter(get_dma_ptr(substream->runtime, channel, hwoff),
- bytes, iter))
+ if (copy_from_iter(get_dma_ptr(substream->runtime, channel, hwoff),
+ bytes, iter) != bytes)
return -EFAULT;
return 0;
}
@@ -2025,8 +2025,8 @@ static int default_read_copy(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
struct iov_iter *iter, unsigned long bytes)
{
- if (!copy_to_iter(get_dma_ptr(substream->runtime, channel, hwoff),
- bytes, iter))
+ if (copy_to_iter(get_dma_ptr(substream->runtime, channel, hwoff),
+ bytes, iter) != bytes)
return -EFAULT;
return 0;
}
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 174585bf59d2..b603bb93f896 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -187,8 +187,13 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char
err = expand_var_event(event, 0, len, buf, in_kernel);
if (err < 0)
return err;
- if (len != newlen)
- memset(buf + len, 0, newlen - len);
+ if (len != newlen) {
+ if (in_kernel)
+ memset(buf + len, 0, newlen - len);
+ else if (clear_user((__force void __user *)buf + len,
+ newlen - len))
+ return -EFAULT;
+ }
return newlen;
}
EXPORT_SYMBOL(snd_seq_expand_var_event);
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
index c05935c2edc4..9234d4fe8ada 100644
--- a/sound/isa/sb/emu8000_pcm.c
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -456,7 +456,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
/* convert to word unit */
pos = (pos << 1) + rec->loop_start[voice];
count <<= 1;
- LOOP_WRITE(rec, pos, USER_SOCKPTR(NULL), count);
+ LOOP_WRITE(rec, pos, NULL, count);
return 0;
}
diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c
index 0ba1fbcbb21e..627899959ffe 100644
--- a/sound/pci/hda/patch_cs8409.c
+++ b/sound/pci/hda/patch_cs8409.c
@@ -888,7 +888,7 @@ static void cs42l42_resume(struct sub_codec *cs42l42)
/* Initialize CS42L42 companion codec */
cs8409_i2c_bulk_write(cs42l42, cs42l42->init_seq, cs42l42->init_seq_num);
- usleep_range(30000, 35000);
+ msleep(CS42L42_INIT_TIMEOUT_MS);
/* Clear interrupts, by reading interrupt status registers */
cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs));
diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h
index 2a8dfb4ff046..937e9387abdc 100644
--- a/sound/pci/hda/patch_cs8409.h
+++ b/sound/pci/hda/patch_cs8409.h
@@ -229,6 +229,7 @@ enum cs8409_coefficient_index_registers {
#define CS42L42_I2C_SLEEP_US (2000)
#define CS42L42_PDN_TIMEOUT_US (250000)
#define CS42L42_PDN_SLEEP_US (2000)
+#define CS42L42_INIT_TIMEOUT_MS (45)
#define CS42L42_FULL_SCALE_VOL_MASK (2)
#define CS42L42_FULL_SCALE_VOL_0DB (1)
#define CS42L42_FULL_SCALE_VOL_MINUS6DB (0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a07df6f92960..b7e78bfcffd8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7057,6 +7057,27 @@ static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec,
}
}
+/* Forcibly assign NID 0x03 to HP while NID 0x02 to SPK */
+static void alc287_fixup_bind_dacs(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+ static const hda_nid_t preferred_pairs[] = {
+ 0x17, 0x02, 0x21, 0x03, 0
+ };
+
+ if (action != HDA_FIXUP_ACT_PRE_PROBE)
+ return;
+
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ spec->gen.preferred_dacs = preferred_pairs;
+ spec->gen.auto_mute_via_amp = 1;
+ snd_hda_codec_write_cache(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ 0x0); /* Make sure 0x14 was disable */
+}
+
+
enum {
ALC269_FIXUP_GPIO2,
ALC269_FIXUP_SONY_VAIO,
@@ -7319,6 +7340,7 @@ enum {
ALC287_FIXUP_TAS2781_I2C,
ALC245_FIXUP_HP_MUTE_LED_COEFBIT,
ALC245_FIXUP_HP_X360_MUTE_LEDS,
+ ALC287_FIXUP_THINKPAD_I2S_SPK,
};
/* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -9413,6 +9435,10 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC245_FIXUP_HP_GPIO_LED
},
+ [ALC287_FIXUP_THINKPAD_I2S_SPK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc287_fixup_bind_dacs,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -10544,6 +10570,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x17, 0x90170111},
{0x19, 0x03a11030},
{0x21, 0x03211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC287_FIXUP_THINKPAD_I2S_SPK,
+ {0x17, 0x90170110},
+ {0x19, 0x03a11030},
+ {0x21, 0x03211020}),
SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
{0x12, 0x90a60130},
{0x17, 0x90170110},
diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c
index 37114fd61a38..fb802802939e 100644
--- a/sound/pci/hda/tas2781_hda_i2c.c
+++ b/sound/pci/hda/tas2781_hda_i2c.c
@@ -173,16 +173,6 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol,
return 0;
}
-static int tasdevice_hda_clamp(int val, int max)
-{
- if (val > max)
- val = max;
-
- if (val < 0)
- val = 0;
- return val;
-}
-
static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -191,7 +181,7 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol,
int max = tas_priv->rcabin.ncfgs - 1;
int val, ret = 0;
- val = tasdevice_hda_clamp(nr_profile, max);
+ val = clamp(nr_profile, 0, max);
if (tas_priv->rcabin.profile_cfg_id != val) {
tas_priv->rcabin.profile_cfg_id = val;
@@ -248,7 +238,7 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol,
int max = tas_fw->nr_programs - 1;
int val, ret = 0;
- val = tasdevice_hda_clamp(nr_program, max);
+ val = clamp(nr_program, 0, max);
if (tas_priv->cur_prog != val) {
tas_priv->cur_prog = val;
@@ -277,7 +267,7 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol,
int max = tas_fw->nr_configurations - 1;
int val, ret = 0;
- val = tasdevice_hda_clamp(nr_config, max);
+ val = clamp(nr_config, 0, max);
if (tas_priv->cur_conf != val) {
tas_priv->cur_conf = val;
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index b304b3562c82..3ec15b46fa35 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -217,6 +217,13 @@ 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, "82TL"),
+ }
+ },
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_NAME, "82V2"),
}
},
@@ -328,6 +335,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
{
.driver_data = &acp6x_card,
.matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
+ DMI_MATCH(DMI_BOARD_NAME, "8A3E"),
+ }
+ },
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "MECHREVO"),
DMI_MATCH(DMI_BOARD_NAME, "MRID6"),
}
diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c
index afe213a71212..dcc4e14b3dde 100644
--- a/sound/soc/atmel/mchp-pdmc.c
+++ b/sound/soc/atmel/mchp-pdmc.c
@@ -954,7 +954,7 @@ static int mchp_pdmc_dt_init(struct mchp_pdmc *dd)
/* used to clean the channel index found on RHR's MSB */
static int mchp_pdmc_process(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- struct iov_iter *buf, unsigned long bytes)
+ unsigned long bytes)
{
struct snd_pcm_runtime *runtime = substream->runtime;
u8 *dma_ptr = runtime->dma_area + hwoff +
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 95b5bd883215..f1e1dbc509f6 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1968,11 +1968,15 @@ config SND_SOC_UDA1380
tristate
depends on I2C
+config SND_SOC_WCD_CLASSH
+ tristate
+
config SND_SOC_WCD9335
tristate "WCD9335 Codec"
depends on SLIMBUS
select REGMAP_SLIMBUS
select REGMAP_IRQ
+ select SND_SOC_WCD_CLASSH
help
The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports
Qualcomm Technologies, Inc. (QTI) multimedia solutions,
@@ -1987,6 +1991,7 @@ config SND_SOC_WCD934X
depends on SLIMBUS
select REGMAP_IRQ
select REGMAP_SLIMBUS
+ select SND_SOC_WCD_CLASSH
select SND_SOC_WCD_MBHC
depends on MFD_WCD934X || COMPILE_TEST
help
@@ -1997,6 +2002,7 @@ config SND_SOC_WCD938X
depends on SND_SOC_WCD938X_SDW
tristate
depends on SOUNDWIRE || !SOUNDWIRE
+ select SND_SOC_WCD_CLASSH
config SND_SOC_WCD938X_SDW
tristate "WCD9380/WCD9385 Codec - SDW"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c8502a49b40a..a87e56938ce5 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -303,10 +303,11 @@ snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
snd-soc-uda1334-objs := uda1334.o
snd-soc-uda1380-objs := uda1380.o
+snd-soc-wcd-classh-objs := wcd-clsh-v2.o
snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o
-snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o
-snd-soc-wcd934x-objs := wcd-clsh-v2.o wcd934x.o
-snd-soc-wcd938x-objs := wcd938x.o wcd-clsh-v2.o
+snd-soc-wcd9335-objs := wcd9335.o
+snd-soc-wcd934x-objs := wcd934x.o
+snd-soc-wcd938x-objs := wcd938x.o
snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o
snd-soc-wl1273-objs := wl1273.o
snd-soc-wm-adsp-objs := wm_adsp.o
@@ -685,6 +686,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
obj-$(CONFIG_SND_SOC_UDA1334) += snd-soc-uda1334.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WCD_CLASSH) += snd-soc-wcd-classh.o
obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o
obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o
obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o
diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c
index d1edb9876c10..be4f4229576c 100644
--- a/sound/soc/codecs/cs35l45.c
+++ b/sound/soc/codecs/cs35l45.c
@@ -279,7 +279,7 @@ static const struct snd_kcontrol_new cs35l45_dsp_muxes[] = {
};
static const struct snd_kcontrol_new cs35l45_dac_muxes[] = {
- SOC_DAPM_ENUM("DACPCM1 Source", cs35l45_dacpcm_enums[0]),
+ SOC_DAPM_ENUM("DACPCM Source", cs35l45_dacpcm_enums[0]),
};
static const struct snd_soc_dapm_widget cs35l45_dapm_widgets[] = {
@@ -333,7 +333,7 @@ static const struct snd_soc_dapm_widget cs35l45_dapm_widgets[] = {
SND_SOC_DAPM_MUX("DSP_RX7 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dsp_muxes[6]),
SND_SOC_DAPM_MUX("DSP_RX8 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dsp_muxes[7]),
- SND_SOC_DAPM_MUX("DACPCM1 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dac_muxes[0]),
+ SND_SOC_DAPM_MUX("DACPCM Source", SND_SOC_NOPM, 0, 0, &cs35l45_dac_muxes[0]),
SND_SOC_DAPM_OUT_DRV("AMP", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -403,7 +403,7 @@ static const struct snd_soc_dapm_route cs35l45_dapm_routes[] = {
{ "ASP_RX1", NULL, "ASP_EN" },
{ "ASP_RX2", NULL, "ASP_EN" },
- { "AMP", NULL, "DACPCM1 Source"},
+ { "AMP", NULL, "DACPCM Source"},
{ "AMP", NULL, "GLOBAL_EN"},
CS35L45_DSP_MUX_ROUTE("DSP_RX1"),
@@ -427,7 +427,7 @@ static const struct snd_soc_dapm_route cs35l45_dapm_routes[] = {
{"DSP1 Preload", NULL, "DSP1 Preloader"},
{"DSP1", NULL, "DSP1 Preloader"},
- CS35L45_DAC_MUX_ROUTE("DACPCM1"),
+ CS35L45_DAC_MUX_ROUTE("DACPCM"),
{ "SPK", NULL, "AMP"},
};
@@ -969,7 +969,7 @@ static irqreturn_t cs35l45_dsp_virt2_mbox_cb(int irq, void *data)
ret = regmap_read(cs35l45->regmap, CS35L45_DSP_VIRT2_MBOX_3, &mbox_val);
if (!ret && mbox_val)
- ret = cs35l45_dsp_virt2_mbox3_irq_handle(cs35l45, mbox_val & CS35L45_MBOX3_CMD_MASK,
+ cs35l45_dsp_virt2_mbox3_irq_handle(cs35l45, mbox_val & CS35L45_MBOX3_CMD_MASK,
(mbox_val & CS35L45_MBOX3_DATA_MASK) >> CS35L45_MBOX3_DATA_SHIFT);
/* Handle DSP trace log IRQ */
@@ -1078,6 +1078,7 @@ static int cs35l45_initialize(struct cs35l45_private *cs35l45)
switch (dev_id[0]) {
case 0x35A450:
+ case 0x35A460:
break;
default:
dev_err(cs35l45->dev, "Bad DEVID 0x%x\n", dev_id[0]);
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index ae373f335ea8..98b1e63360ae 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -243,26 +243,27 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base)
{
unsigned int reg;
unsigned int val;
- int ret;
+ int read_ret, poll_ret;
if (cs35l56_base->rev < CS35L56_REVID_B0)
reg = CS35L56_DSP1_HALO_STATE_A1;
else
reg = CS35L56_DSP1_HALO_STATE;
- ret = regmap_read_poll_timeout(cs35l56_base->regmap, reg,
- val,
- (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE),
- CS35L56_HALO_STATE_POLL_US,
- CS35L56_HALO_STATE_TIMEOUT_US);
-
- if ((ret < 0) && (ret != -ETIMEDOUT)) {
- dev_err(cs35l56_base->dev, "Failed to read HALO_STATE: %d\n", ret);
- return ret;
- }
-
- if ((ret == -ETIMEDOUT) || (val != CS35L56_HALO_STATE_BOOT_DONE)) {
- dev_err(cs35l56_base->dev, "Firmware boot fail: HALO_STATE=%#x\n", val);
+ /*
+ * This can't be a regmap_read_poll_timeout() because cs35l56 will NAK
+ * I2C until it has booted which would terminate the poll
+ */
+ poll_ret = read_poll_timeout(regmap_read, read_ret,
+ (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE),
+ CS35L56_HALO_STATE_POLL_US,
+ CS35L56_HALO_STATE_TIMEOUT_US,
+ false,
+ cs35l56_base->regmap, reg, &val);
+
+ if (poll_ret) {
+ dev_err(cs35l56_base->dev, "Firmware boot timed out(%d): HALO_STATE=%#x\n",
+ read_ret, val);
return -EIO;
}
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 24e718e51174..1a95c370fc4c 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2205,7 +2205,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev)
// Don't use devm as we need to get against the MFD device
priv->mclk = clk_get_optional(cs42l43->dev, "mclk");
if (IS_ERR(priv->mclk)) {
- dev_err_probe(priv->dev, PTR_ERR(priv->mclk), "Failed to get mclk\n");
+ ret = PTR_ERR(priv->mclk);
+ dev_err_probe(priv->dev, ret, "Failed to get mclk\n");
goto err_pm;
}
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 038d93e20883..1a137ca3f496 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3269,13 +3269,17 @@ static int rt5645_component_set_jack(struct snd_soc_component *component,
{
struct snd_soc_jack *mic_jack = NULL;
struct snd_soc_jack *btn_jack = NULL;
- int *type = (int *)data;
+ int type;
- if (*type & SND_JACK_MICROPHONE)
- mic_jack = hs_jack;
- if (*type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
- SND_JACK_BTN_2 | SND_JACK_BTN_3))
- btn_jack = hs_jack;
+ if (hs_jack) {
+ type = *(int *)data;
+
+ if (type & SND_JACK_MICROPHONE)
+ mic_jack = hs_jack;
+ if (type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3))
+ btn_jack = hs_jack;
+ }
return rt5645_set_jack_detect(component, hs_jack, mic_jack, btn_jack);
}
diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c
index a75db27e5205..d96e23ec43d4 100644
--- a/sound/soc/codecs/wcd-clsh-v2.c
+++ b/sound/soc/codecs/wcd-clsh-v2.c
@@ -355,6 +355,7 @@ void wcd_clsh_set_hph_mode(struct wcd_clsh_ctrl *ctrl, int mode)
wcd_clsh_v2_set_hph_mode(comp, mode);
}
+EXPORT_SYMBOL_GPL(wcd_clsh_set_hph_mode);
static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp,
int mode)
@@ -869,11 +870,13 @@ int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl,
return 0;
}
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_set_state);
int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl)
{
return ctrl->state;
}
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_get_state);
struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp,
int version)
@@ -890,8 +893,13 @@ struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp,
return ctrl;
}
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_alloc);
void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl)
{
kfree(ctrl);
}
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_free);
+
+MODULE_DESCRIPTION("WCD93XX Class-H driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c
index 1fbb2c2fadb5..8565a530706d 100644
--- a/sound/soc/intel/avs/pcm.c
+++ b/sound/soc/intel/avs/pcm.c
@@ -796,6 +796,28 @@ static int avs_component_probe(struct snd_soc_component *component)
ret = avs_load_topology(component, filename);
kfree(filename);
+ if (ret == -ENOENT && !strncmp(mach->tplg_filename, "hda-", 4)) {
+ unsigned int vendor_id;
+
+ if (sscanf(mach->tplg_filename, "hda-%08x-tplg.bin", &vendor_id) != 1)
+ return ret;
+
+ if (((vendor_id >> 16) & 0xFFFF) == 0x8086)
+ mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL,
+ "hda-8086-generic-tplg.bin");
+ else
+ mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL,
+ "hda-generic-tplg.bin");
+
+ filename = kasprintf(GFP_KERNEL, "%s/%s", component->driver->topology_name_prefix,
+ mach->tplg_filename);
+ if (!filename)
+ return -ENOMEM;
+
+ dev_info(card->dev, "trying to load fallback topology %s\n", mach->tplg_filename);
+ ret = avs_load_topology(component, filename);
+ kfree(filename);
+ }
if (ret < 0)
return ret;
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index f18406dfa1e4..ba7c0ae82e00 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -1054,7 +1054,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream)
int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
int channel, unsigned long pos,
- struct iov_iter *buf, unsigned long bytes)
+ struct iov_iter *iter, unsigned long bytes)
{
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_component *component;
@@ -1065,7 +1065,7 @@ int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
if (component->driver->copy)
return soc_component_ret(component,
component->driver->copy(component, substream,
- channel, pos, buf, bytes));
+ channel, pos, iter, bytes));
return -EINVAL;
}
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index ff2166525dbc..d0653d775c87 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -290,29 +290,29 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer(
static int dmaengine_copy(struct snd_soc_component *component,
struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- struct iov_iter *buf, unsigned long bytes)
+ struct iov_iter *iter, unsigned long bytes)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct dmaengine_pcm *pcm = soc_component_to_pcm(component);
int (*process)(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- struct iov_iter *buf, unsigned long bytes) = pcm->config->process;
+ unsigned long bytes) = pcm->config->process;
bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
void *dma_ptr = runtime->dma_area + hwoff +
channel * (runtime->dma_bytes / runtime->channels);
if (is_playback)
- if (copy_from_iter(dma_ptr, bytes, buf) != bytes)
+ if (copy_from_iter(dma_ptr, bytes, iter) != bytes)
return -EFAULT;
if (process) {
- int ret = process(substream, channel, hwoff, buf, bytes);
+ int ret = process(substream, channel, hwoff, bytes);
if (ret < 0)
return ret;
}
if (!is_playback)
- if (copy_to_iter(dma_ptr, bytes, buf) != bytes)
+ if (copy_to_iter(dma_ptr, bytes, iter) != bytes)
return -EFAULT;
return 0;
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index f9b5d5969155..0acc848c1f00 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1246,7 +1246,7 @@ static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops2 = {
static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
int channel, unsigned long hwoff,
- struct iov_iter *buf, unsigned long bytes)
+ unsigned long bytes)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c
index a27e244650c8..1ec177fe284e 100644
--- a/sound/usb/midi2.c
+++ b/sound/usb/midi2.c
@@ -265,7 +265,7 @@ static void free_midi_urbs(struct snd_usb_midi2_endpoint *ep)
if (!ep)
return;
- for (i = 0; i < ep->num_urbs; ++i) {
+ for (i = 0; i < NUM_URBS; ++i) {
ctx = &ep->urbs[i];
if (!ctx->urb)
break;
@@ -279,6 +279,7 @@ static void free_midi_urbs(struct snd_usb_midi2_endpoint *ep)
}
/* allocate URBs for an EP */
+/* the callers should handle allocation errors via free_midi_urbs() */
static int alloc_midi_urbs(struct snd_usb_midi2_endpoint *ep)
{
struct snd_usb_midi2_urb *ctx;
@@ -351,8 +352,10 @@ static int snd_usb_midi_v2_open(struct snd_ump_endpoint *ump, int dir)
return -EIO;
if (ep->direction == STR_OUT) {
err = alloc_midi_urbs(ep);
- if (err)
+ if (err) {
+ free_midi_urbs(ep);
return err;
+ }
}
return 0;
}