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Subject[GIT PULL] sound fixes for 5.0-rc7
Linus,

please pull sound fixes for v5.0-rc7 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-5.0-rc7

The topmost commit is 00a399cad1a063e7665f06b6497a807db20441fd

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

sound fixes for 5.0-rc7

It's a bit of surprising that we've got more changes than hoped
at this late stage, but the all don't look too scaring but small
fixes.

One change in ALSA core side is again the PCM regression fix that
was partially addressed for OSS, but now the all relevant change
is reverted instead. Also, a few ASoC core fixes for UAF and OOB
are included, while the rest are usual random device-specific
fixes.

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

Charles Keepax (1):
ASoC: core: Allow soc_find_component lookups to match parent of_node

Fabio Estevam (1):
ASoC: MAINTAINERS: fsl: Change Fabio's email address

Guennadi Liakhovetski (1):
ASoC: topology: fix oops/use-after-free case with dai driver

Jiada Wang (1):
ASoC: rsnd: ssiu: correct shift bit for ssiu9

Jurica Vukadin (1):
ALSA: hda - Add quirk for HP EliteBook 840 G5

Kuninori Morimoto (2):
ASoC: rsnd: fixup MIX kctrl registration
ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check

Manuel Reinhardt (1):
ALSA: usb-audio: Fix implicit fb endpoint setup by quirk

Pierre-Louis Bossart (1):
ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables

Russell King (1):
ASoC: hdmi-codec: fix oops on re-probe

Shuming Fan (1):
ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter

Sylwester Nawrocki (1):
ASoC: samsung: Prevent clk_get_rate() calls in atomic context

Takashi Iwai (1):
ALSA: pcm: Revert capture stream behavior change in blocking mode

---
MAINTAINERS | 4 ++--
sound/core/pcm_lib.c | 20 ++++----------------
sound/pci/hda/patch_conexant.c | 1 +
sound/soc/codecs/hdmi-codec.c | 4 ++--
sound/soc/codecs/rt5682.c | 2 ++
sound/soc/samsung/i2s.c | 10 +++++-----
sound/soc/sh/rcar/core.c | 8 ++++----
sound/soc/sh/rcar/ssi.c | 2 +-
sound/soc/sh/rcar/ssiu.c | 2 +-
sound/soc/soc-core.c | 9 +++++++--
sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++
sound/soc/soc-topology.c | 5 +++++
sound/usb/pcm.c | 9 ++++++++-
13 files changed, 66 insertions(+), 34 deletions(-)

diff --git a/MAINTAINERS b/MAINTAINERS
index 9919840d54cd..41ce5f4ad838 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6151,7 +6151,7 @@ FREESCALE SOC SOUND DRIVERS
M: Timur Tabi <timur@kernel.org>
M: Nicolin Chen <nicoleotsuka@gmail.com>
M: Xiubo Li <Xiubo.Lee@gmail.com>
-R: Fabio Estevam <fabio.estevam@nxp.com>
+R: Fabio Estevam <festevam@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: linuxppc-dev@lists.ozlabs.org
S: Maintained
@@ -10898,7 +10898,7 @@ F: include/linux/nvmem-consumer.h
F: include/linux/nvmem-provider.h

NXP SGTL5000 DRIVER
-M: Fabio Estevam <fabio.estevam@nxp.com>
+M: Fabio Estevam <festevam@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Maintained
F: Documentation/devicetree/bindings/sound/sgtl5000.txt
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6c99fa8ac5fa..6c0b30391ba9 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2112,13 +2112,6 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream,
return 0;
}

-/* allow waiting for a capture stream that hasn't been started */
-#if IS_ENABLED(CONFIG_SND_PCM_OSS)
-#define wait_capture_start(substream) ((substream)->oss.oss)
-#else
-#define wait_capture_start(substream) false
-#endif
-
/* the common loop for read/write data */
snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
void *data, bool interleaved,
@@ -2184,16 +2177,11 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
snd_pcm_update_hw_ptr(substream);

if (!is_playback &&
- runtime->status->state == SNDRV_PCM_STATE_PREPARED) {
- if (size >= runtime->start_threshold) {
- err = snd_pcm_start(substream);
- if (err < 0)
- goto _end_unlock;
- } else if (!wait_capture_start(substream)) {
- /* nothing to do */
- err = 0;
+ runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
+ size >= runtime->start_threshold) {
+ err = snd_pcm_start(substream);
+ if (err < 0)
goto _end_unlock;
- }
}

avail = snd_pcm_avail(substream);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 152f54137082..a4ee7656d9ee 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -924,6 +924,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+ SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index d00734d31e04..e5b6769b9797 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -795,6 +795,8 @@ static int hdmi_codec_probe(struct platform_device *pdev)
if (hcd->spdif)
hcp->daidrv[i] = hdmi_spdif_dai;

+ dev_set_drvdata(dev, hcp);
+
ret = devm_snd_soc_register_component(dev, &hdmi_driver, hcp->daidrv,
dai_count);
if (ret) {
@@ -802,8 +804,6 @@ static int hdmi_codec_probe(struct platform_device *pdev)
__func__, ret);
return ret;
}
-
- dev_set_drvdata(dev, hcp);
return 0;
}

diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 89c43b26c379..a9b91bcfcc09 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -1778,7 +1778,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
{"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
{"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc},
{"ADC STO1 ASRC", NULL, "AD ASRC"},
+ {"ADC STO1 ASRC", NULL, "DA ASRC"},
{"ADC STO1 ASRC", NULL, "CLKDET"},
+ {"DAC STO1 ASRC", NULL, "AD ASRC"},
{"DAC STO1 ASRC", NULL, "DA ASRC"},
{"DAC STO1 ASRC", NULL, "CLKDET"},

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index d6c62aa13041..ce00fe2f6aae 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -700,6 +700,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
{
struct i2s_dai *i2s = to_info(dai);
u32 mod, mask = 0, val = 0;
+ struct clk *rclksrc;
unsigned long flags;

WARN_ON(!pm_runtime_active(dai->dev));
@@ -782,6 +783,10 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,

i2s->frmclk = params_rate(params);

+ rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC];
+ if (rclksrc && !IS_ERR(rclksrc))
+ i2s->rclk_srcrate = clk_get_rate(rclksrc);
+
return 0;
}

@@ -886,11 +891,6 @@ static int config_setup(struct i2s_dai *i2s)
return 0;

if (!(i2s->quirks & QUIRK_NO_MUXPSR)) {
- struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC];
-
- if (rclksrc && !IS_ERR(rclksrc))
- i2s->rclk_srcrate = clk_get_rate(rclksrc);
-
psr = i2s->rclk_srcrate / i2s->frmclk / rfs;
writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR);
dev_dbg(&i2s->pdev->dev,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 59e250cc2e9d..e819e965e1db 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1526,14 +1526,14 @@ int rsnd_kctrl_new(struct rsnd_mod *mod,
int ret;

/*
- * 1) Avoid duplicate register (ex. MIXer case)
- * 2) re-register if card was rebinded
+ * 1) Avoid duplicate register for DVC with MIX case
+ * 2) Allow duplicate register for MIX
+ * 3) re-register if card was rebinded
*/
list_for_each_entry(kctrl, &card->controls, list) {
struct rsnd_kctrl_cfg *c = kctrl->private_data;

- if (strcmp(kctrl->id.name, name) == 0 &&
- c->mod == mod)
+ if (c == cfg)
return 0;
}

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 45ef295743ec..f5afab631abb 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
if (rsnd_ssi_is_multi_slave(mod, io))
return 0;

- if (ssi->usrcnt > 1) {
+ if (ssi->usrcnt > 0) {
if (ssi->rate != rate) {
dev_err(dev, "SSI parent/child should use same rate\n");
return -EINVAL;
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index c5934adcfd01..c74991dd18ab 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -79,7 +79,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod,
break;
case 9:
for (i = 0; i < 4; i++)
- rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4));
+ rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4);
break;
}

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index aae450ba4f08..50617db05c46 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -735,12 +735,17 @@ static struct snd_soc_component *soc_find_component(
const struct device_node *of_node, const char *name)
{
struct snd_soc_component *component;
+ struct device_node *component_of_node;

lockdep_assert_held(&client_mutex);

for_each_component(component) {
if (of_node) {
- if (component->dev->of_node == of_node)
+ component_of_node = component->dev->of_node;
+ if (!component_of_node && component->dev->parent)
+ component_of_node = component->dev->parent->of_node;
+
+ if (component_of_node == of_node)
return component;
} else if (name && strcmp(component->name, name) == 0) {
return component;
@@ -951,7 +956,7 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
{
int err;

- if (!dai || !dai->probed ||
+ if (!dai || !dai->probed || !dai->driver ||
dai->driver->remove_order != order)
return;

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2c4c13419539..20bad755888b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -70,12 +70,16 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_clock_supply] = 1,
[snd_soc_dapm_supply] = 2,
[snd_soc_dapm_micbias] = 3,
+ [snd_soc_dapm_vmid] = 3,
[snd_soc_dapm_dai_link] = 2,
[snd_soc_dapm_dai_in] = 4,
[snd_soc_dapm_dai_out] = 4,
[snd_soc_dapm_aif_in] = 4,
[snd_soc_dapm_aif_out] = 4,
[snd_soc_dapm_mic] = 5,
+ [snd_soc_dapm_siggen] = 5,
+ [snd_soc_dapm_input] = 5,
+ [snd_soc_dapm_output] = 5,
[snd_soc_dapm_mux] = 6,
[snd_soc_dapm_demux] = 6,
[snd_soc_dapm_dac] = 7,
@@ -83,11 +87,19 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_mixer] = 8,
[snd_soc_dapm_mixer_named_ctl] = 8,
[snd_soc_dapm_pga] = 9,
+ [snd_soc_dapm_buffer] = 9,
+ [snd_soc_dapm_scheduler] = 9,
+ [snd_soc_dapm_effect] = 9,
+ [snd_soc_dapm_src] = 9,
+ [snd_soc_dapm_asrc] = 9,
+ [snd_soc_dapm_encoder] = 9,
+ [snd_soc_dapm_decoder] = 9,
[snd_soc_dapm_adc] = 10,
[snd_soc_dapm_out_drv] = 11,
[snd_soc_dapm_hp] = 11,
[snd_soc_dapm_spk] = 11,
[snd_soc_dapm_line] = 11,
+ [snd_soc_dapm_sink] = 11,
[snd_soc_dapm_kcontrol] = 12,
[snd_soc_dapm_post] = 13,
};
@@ -100,13 +112,25 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_spk] = 3,
[snd_soc_dapm_line] = 3,
[snd_soc_dapm_out_drv] = 3,
+ [snd_soc_dapm_sink] = 3,
[snd_soc_dapm_pga] = 4,
+ [snd_soc_dapm_buffer] = 4,
+ [snd_soc_dapm_scheduler] = 4,
+ [snd_soc_dapm_effect] = 4,
+ [snd_soc_dapm_src] = 4,
+ [snd_soc_dapm_asrc] = 4,
+ [snd_soc_dapm_encoder] = 4,
+ [snd_soc_dapm_decoder] = 4,
[snd_soc_dapm_switch] = 5,
[snd_soc_dapm_mixer_named_ctl] = 5,
[snd_soc_dapm_mixer] = 5,
[snd_soc_dapm_dac] = 6,
[snd_soc_dapm_mic] = 7,
+ [snd_soc_dapm_siggen] = 7,
+ [snd_soc_dapm_input] = 7,
+ [snd_soc_dapm_output] = 7,
[snd_soc_dapm_micbias] = 8,
+ [snd_soc_dapm_vmid] = 8,
[snd_soc_dapm_mux] = 9,
[snd_soc_dapm_demux] = 9,
[snd_soc_dapm_aif_in] = 10,
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 045ef136903d..fc79ec6927e3 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp,
{
struct snd_soc_dai_driver *dai_drv =
container_of(dobj, struct snd_soc_dai_driver, dobj);
+ struct snd_soc_dai *dai;

if (pass != SOC_TPLG_PASS_PCM_DAI)
return;
@@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp,
if (dobj->ops && dobj->ops->dai_unload)
dobj->ops->dai_unload(comp, dobj);

+ list_for_each_entry(dai, &comp->dai_list, list)
+ if (dai->driver == dai_drv)
+ dai->driver = NULL;
+
kfree(dai_drv->name);
list_del(&dobj->list);
kfree(dai_drv);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 382847154227..db114f3977e0 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
return 0;
}

+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
struct usb_device *dev,
struct usb_interface_descriptor *altsd,
@@ -384,7 +387,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,

subs->data_endpoint->sync_master = subs->sync_endpoint;

- return 0;
+ return 1;
}

static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
if (err < 0)
return err;

+ /* endpoint set by quirk */
+ if (err > 0)
+ return 0;
+
if (altsd->bNumEndpoints < 2)
return 0;

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