From cbb9f8ccc8232b4647c4180af653eee744818221 Mon Sep 17 00:00:00 2001
From: John Hsu <KCHSU0@nuvoton.com>
Date: Mon, 10 Jun 2019 11:40:40 +0800
Subject: [PATCH 01/14] ASoC: nau8825: fix fake interruption when booting
There is no pull-up resistor at IRQ line where it connects from
the codec to SoC. When booting, the signal of IRQ pin will keep low
which makes the SoC invoke the ISR repeatedly because the IRQ is
registered trigger low. It will not stop until the codec sets up
the interruption and pulls the signal high. In the patch,
nau8825 will internally pull the signal to high at booting in case
the fake interrupts happen.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Tested-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/nau8825.c | 4 ++++
sound/soc/codecs/nau8825.h | 2 ++
2 files changed, 6 insertions(+)
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 47e65cf99879..83ec841f7865 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -1881,6 +1881,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825)
NAU8825_JACK_EJECT_DEBOUNCE_MASK,
nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT);
+ /* Pull up IRQ pin */
+ regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK,
+ NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN,
+ NAU8825_IRQ_PIN_PULLUP | NAU8825_IRQ_PIN_PULL_EN);
/* Mask unneeded IRQs: 1 - disable, 0 - enable */
regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff);
diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h
index f6074c618569..3f41897ed3f6 100644
--- a/sound/soc/codecs/nau8825.h
+++ b/sound/soc/codecs/nau8825.h
@@ -171,6 +171,8 @@
#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */
/* INTERRUPT_MASK (0xf) */
+#define NAU8825_IRQ_PIN_PULLUP (1 << 14)
+#define NAU8825_IRQ_PIN_PULL_EN (1 << 13)
#define NAU8825_IRQ_OUTPUT_EN (1 << 11)
#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10)
#define NAU8825_IRQ_RMS_EN (1 << 8)
From 47c317b786b6c1efc2cb3cdb894fd323422fe5ea Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 13 Jun 2019 13:42:30 +0200
Subject: [PATCH 02/14] ASoC: meson: axg-tdmin: right_j is not supported
Right justified format is actually not supported by the amlogic tdm input
decoder.
Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/axg-tdmin.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index a790f925a4ef..cb87f17f3e95 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -121,7 +121,6 @@ static int axg_tdmin_prepare(struct regmap *map,
break;
case SND_SOC_DAIFMT_LEFT_J:
- case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_DSP_B:
break;
From 7e0d7d0fbd06af0507611f85dba8daf24832abd9 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 13 Jun 2019 13:42:31 +0200
Subject: [PATCH 03/14] ASoC: meson: axg-tdmout: right_j is not supported
Right justified format is actually not supported by the amlogic tdm output
encoder.
Fixes: c41c2a355b86 ("ASoC: meson: add tdm output driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/axg-tdmout.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 527bfc4487e0..86537fc0ecb5 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -137,7 +137,6 @@ static int axg_tdmout_prepare(struct regmap *map,
break;
case SND_SOC_DAIFMT_LEFT_J:
- case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_DSP_B:
skew += 1;
break;
From cb36ff785e868992e96e8b9e5a0c2822b680a9e2 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Thu, 13 Jun 2019 13:42:32 +0200
Subject: [PATCH 04/14] ASoC: meson: axg-tdm: fix sample clock inversion
The content of SND_SOC_DAIFMT_FORMAT_MASK is a number, not a bitfield,
so the test to check if the format is i2s is wrong. Because of this the
clock setting may be wrong. For example, the sample clock gets inverted
in DSP B mode, when it should not.
Fix the lrclk invert helper function
Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/meson/axg-tdm.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h
index e578b6f40a07..5774ce0916d4 100644
--- a/sound/soc/meson/axg-tdm.h
+++ b/sound/soc/meson/axg-tdm.h
@@ -40,7 +40,7 @@ struct axg_tdm_iface {
static inline bool axg_tdm_lrclk_invert(unsigned int fmt)
{
- return (fmt & SND_SOC_DAIFMT_I2S) ^
+ return ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) ^
!!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF));
}
From 489f231e0f4c44d4d019aa5c26e1c3f147875f13 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 13 Jun 2019 14:54:12 +0300
Subject: [PATCH 05/14] ASoC: codec: hdac_hdmi: fix pin connections at cvt
enable
In display codecs supported by hdac_hdmi, the connection indices are
shared by all converters. At boot and resume from suspend,
the connection state may be reset to default values.
In case of multiple connected pins (multiple monitors connected
with audio capability), routing and mute status of pins that
are not connected to any PCM, may interfere with other pins.
E.g. after resume from S3 with multiple monitors, unless
all converters are in active use, playback to some PCMs may
be muted due to the default settings of unrelated converters.
Avoid this by ensuring all pin:cvt selections are correct
in codec whenever a converter is enabled for playback.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/hdac_hdmi.c | 31 +++++++++++++++++++++++++++++++
1 file changed, 31 insertions(+)
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 660e0587f399..7eba57157bb9 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -546,6 +546,29 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt(
return NULL;
}
+/*
+ * Go through all converters and ensure connection is set to
+ * the correct pin as set via kcontrols.
+ */
+static void hdac_hdmi_verify_connect_sel_all_pins(struct hdac_device *hdev)
+{
+ struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+ struct hdac_hdmi_port *port;
+ struct hdac_hdmi_cvt *cvt;
+ int cvt_idx = 0;
+
+ list_for_each_entry(cvt, &hdmi->cvt_list, head) {
+ port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt);
+ if (port && port->pin) {
+ snd_hdac_codec_write(hdev, port->pin->nid, 0,
+ AC_VERB_SET_CONNECT_SEL, cvt_idx);
+ dev_dbg(&hdev->dev, "%s: %s set connect %d -> %d\n",
+ __func__, cvt->name, port->pin->nid, cvt_idx);
+ }
+ ++cvt_idx;
+ }
+}
+
/*
* This tries to get a valid pin and set the HW constraints based on the
* ELD. Even if a valid pin is not found return success so that device open
@@ -806,6 +829,14 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w,
AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag);
snd_hdac_codec_write(hdev, cvt->nid, 0,
AC_VERB_SET_STREAM_FORMAT, pcm->format);
+
+ /*
+ * The connection indices are shared by all converters and
+ * may interfere with each other. Ensure correct
+ * routing for all converters at stream start.
+ */
+ hdac_hdmi_verify_connect_sel_all_pins(hdev);
+
break;
case SND_SOC_DAPM_POST_PMD:
From 3e802e90ffcce333127d928eaefdfcc34af233e8 Mon Sep 17 00:00:00 2001
From: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Date: Sun, 2 Jun 2019 16:55:49 +0200
Subject: [PATCH 06/14] ASoC: ti: Fix SDMA users not providing channel names
McBSP used to work correctly as long as compat DMA probing, removed by
commit 642aafea8889 ("ASoC: ti: remove compat dma probing"), was
available. New method of DMA probing apparently requires users to
provide channel names when registering with SDMA, while McBSP passes
NULLs. Fix it.
The same probably applies to McASP (not tested), hence the patch fixes
both.
Fixes: 642aafea8889 ("ASoC: ti: remove compat dma probing")
Signed-off-by: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-mcasp.c | 2 +-
sound/soc/ti/omap-mcbsp.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 9fbc759fdefe..f31805920e3e 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -2237,7 +2237,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
ret = edma_pcm_platform_register(&pdev->dev);
break;
case PCM_SDMA:
- ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+ ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
break;
default:
dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index a395598f1f20..610c5e706fd2 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -1438,7 +1438,7 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
if (ret)
return ret;
- return sdma_pcm_platform_register(&pdev->dev, NULL, NULL);
+ return sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
}
static int asoc_mcbsp_remove(struct platform_device *pdev)
From 7a1954de3050cb13cf9ee43493ea45785dae68a2 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 13 Jun 2019 21:04:30 +0200
Subject: [PATCH 07/14] ASoC: Intel: Skylake: Fix incorrect capture position
reporting
HW recommends to set DUM bit on device power up, so that DPIB write
request occurs every frame regardless of whether DPIB has changed or
not. This addresses incorrect position reporting for capture streams.
Signed-off-by: Leoni Prodduvaka <leoni.prodduvaka@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/skylake/skl.c | 21 +++++++++++++++++++++
sound/soc/intel/skylake/skl.h | 1 +
2 files changed, 22 insertions(+)
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 4ed5b7e17d44..16f4372ce437 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -192,6 +192,25 @@ void skl_update_d0i3c(struct device *dev, bool enable)
snd_hdac_chip_readb(bus, VS_D0I3C));
}
+/**
+ * skl_dum_set - set DUM bit in EM2 register
+ * @bus: HD-audio core bus
+ *
+ * Addresses incorrect position reporting for capture streams.
+ * Used on device power up.
+ */
+static void skl_dum_set(struct hdac_bus *bus)
+{
+ /* For the DUM bit to be set, CRST needs to be out of reset state */
+ if (!(snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET)) {
+ skl_enable_miscbdcge(bus->dev, false);
+ snd_hdac_bus_exit_link_reset(bus);
+ skl_enable_miscbdcge(bus->dev, true);
+ }
+
+ snd_hdac_chip_updatel(bus, VS_EM2, AZX_VS_EM2_DUM, AZX_VS_EM2_DUM);
+}
+
/* called from IRQ */
static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr)
{
@@ -299,6 +318,7 @@ static int _skl_resume(struct hdac_bus *bus)
struct skl *skl = bus_to_skl(bus);
skl_init_pci(skl);
+ skl_dum_set(bus);
skl_init_chip(bus, true);
return skl_resume_dsp(skl);
@@ -956,6 +976,7 @@ static int skl_first_init(struct hdac_bus *bus)
/* initialize chip */
skl_init_pci(skl);
+ skl_dum_set(bus);
return skl_init_chip(bus, true);
}
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 85f8bb6687dc..b92a7f8fe675 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -46,6 +46,7 @@
#define DMA_TRANSMITION_START 2
#define DMA_TRANSMITION_STOP 3
+#define AZX_VS_EM2_DUM BIT(23)
#define AZX_REG_VS_EM2_L1SEN BIT(13)
struct skl_dsp_resource {
From c054b41690a44e6534eb2a1beda1b655f3994c5b Mon Sep 17 00:00:00 2001
From: Bjorn Andersson <bjorn.andersson@linaro.org>
Date: Mon, 17 Jun 2019 22:29:09 -0700
Subject: [PATCH 08/14] ASoC: qcom: common: Mark links as nonatomic
The interface used to communicate with the DSP can sleep, so mark the
links as nonatomic. This prevents various sleep while atomic errors when
bringing up the audio interface.
Suggested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/common.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 5661025e8cec..a612d860ad26 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -97,6 +97,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
goto err;
}
+ link->nonatomic = 1;
link->dpcm_playback = 1;
link->dpcm_capture = 1;
link->stream_name = link->name;
From 281c443f1e8e25ebc46aaebf48c73e0545a0830e Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Tue, 18 Jun 2019 15:04:26 +0800
Subject: [PATCH 09/14] ASoC: Intel: sof_rt5682: use GFP_KERNEL instead of
GFP_ATOMIC
Change the memory allocation flag from GFP_ATOMIC to GFP_KERNEL because
probe of platform device is unlikely a place where cannot sleep.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_rt5682.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 3343dbcd506f..90d262ebeb10 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -494,7 +494,7 @@ static int sof_audio_probe(struct platform_device *pdev)
int dmic_num, hdmi_num;
int ret, ssp_amp, ssp_codec;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)
return -ENOMEM;
From b545542a0b866f7975254e41c595836e9bc0ff2f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Jun 2019 10:07:19 +0900
Subject: [PATCH 10/14] ASoC: soc-core: call snd_soc_unbind_card() under
mutex_lock;
commit 34ac3c3eb8f0c07 ("ASoC: core: lock client_mutex while removing
link components") added mutex_lock() at soc_remove_link_components().
Is is called from snd_soc_unbind_card()
snd_soc_unbind_card()
=> soc_remove_link_components()
soc_cleanup_card_resources()
soc_remove_dai_links()
=> soc_remove_link_components()
And, there are 2 way to call it.
(1)
snd_soc_unregister_component()
** mutex_lock()
snd_soc_component_del_unlocked()
=> snd_soc_unbind_card()
** mutex_unlock()
(2)
snd_soc_unregister_card()
=> snd_soc_unbind_card()
(1) case is already using mutex_lock() when it calles
snd_soc_unbind_card(), thus, we will get lockdep warning.
commit 495f926c68ddb90 ("ASoC: core: Fix deadlock in
snd_soc_instantiate_card()") tried to fixup it, but still not
enough. We still have lockdep warning when we try unbind/bind.
We need mutex_lock() under snd_soc_unregister_card()
instead of snd_remove_link_components()/snd_soc_unbind_card().
Fixes: 34ac3c3eb8f0c07 ("ASoC: core: lock client_mutex while removing link components")
Fixes: 495f926c68ddb90 ("ASoC: core: Fix deadlock in snd_soc_instantiate_card()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 41c0cfaf2db5..9138fcb15cd3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2837,14 +2837,12 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
snd_soc_dapm_shutdown(card);
snd_soc_flush_all_delayed_work(card);
- mutex_lock(&client_mutex);
/* remove all components used by DAI links on this card */
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
soc_remove_link_components(card, rtd, order);
}
}
- mutex_unlock(&client_mutex);
soc_cleanup_card_resources(card);
if (!unregister)
@@ -2863,7 +2861,9 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
*/
int snd_soc_unregister_card(struct snd_soc_card *card)
{
+ mutex_lock(&client_mutex);
snd_soc_unbind_card(card, true);
+ mutex_unlock(&client_mutex);
dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
return 0;
From c2c928c93173f220955030e8440517b87ec7df92 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Fri, 21 Jun 2019 12:33:56 +0100
Subject: [PATCH 11/14] ASoC: core: Adapt for debugfs API change
Back in ff9fb72bc07705c (debugfs: return error values, not NULL) the
debugfs APIs were changed to return error pointers rather than NULL
pointers on error, breaking the error checking in ASoC. Update the
code to use IS_ERR() and log the codes that are returned as part of
the error messages.
Fixes: ff9fb72bc07705c (debugfs: return error values, not NULL)
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 16 ++++++++++------
1 file changed, 10 insertions(+), 6 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9138fcb15cd3..6aeba0d66ec5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -158,9 +158,10 @@ static void soc_init_component_debugfs(struct snd_soc_component *component)
component->card->debugfs_card_root);
}
- if (!component->debugfs_root) {
+ if (IS_ERR(component->debugfs_root)) {
dev_warn(component->dev,
- "ASoC: Failed to create component debugfs directory\n");
+ "ASoC: Failed to create component debugfs directory: %ld\n",
+ PTR_ERR(component->debugfs_root));
return;
}
@@ -212,18 +213,21 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
card->debugfs_card_root = debugfs_create_dir(card->name,
snd_soc_debugfs_root);
- if (!card->debugfs_card_root) {
+ if (IS_ERR(card->debugfs_card_root)) {
dev_warn(card->dev,
- "ASoC: Failed to create card debugfs directory\n");
+ "ASoC: Failed to create card debugfs directory: %ld\n",
+ PTR_ERR(card->debugfs_card_root));
+ card->debugfs_card_root = NULL;
return;
}
card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
card->debugfs_card_root,
&card->pop_time);
- if (!card->debugfs_pop_time)
+ if (IS_ERR(card->debugfs_pop_time))
dev_warn(card->dev,
- "ASoC: Failed to create pop time debugfs file\n");
+ "ASoC: Failed to create pop time debugfs file: %ld\n",
+ PTR_ERR(card->debugfs_pop_time));
}
static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
From ceaea851b9ea75f9ea2bbefb53ff0d4b27cd5a6e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Fri, 21 Jun 2019 12:33:57 +0100
Subject: [PATCH 12/14] ASoC: dapm: Adapt for debugfs API change
Back in ff9fb72bc07705c (debugfs: return error values, not NULL) the
debugfs APIs were changed to return error pointers rather than NULL
pointers on error, breaking the error checking in ASoC. Update the
code to use IS_ERR() and log the codes that are returned as part of
the error messages.
Fixes: ff9fb72bc07705c (debugfs: return error values, not NULL)
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 18 ++++++++++--------
1 file changed, 10 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5fc57af9cb6f..a248d88b8968 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2154,23 +2154,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
{
struct dentry *d;
- if (!parent)
+ if (!parent || IS_ERR(parent))
return;
dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
- if (!dapm->debugfs_dapm) {
+ if (IS_ERR(dapm->debugfs_dapm)) {
dev_warn(dapm->dev,
- "ASoC: Failed to create DAPM debugfs directory\n");
+ "ASoC: Failed to create DAPM debugfs directory %ld\n",
+ PTR_ERR(dapm->debugfs_dapm));
return;
}
d = debugfs_create_file("bias_level", 0444,
dapm->debugfs_dapm, dapm,
&dapm_bias_fops);
- if (!d)
+ if (IS_ERR(d))
dev_warn(dapm->dev,
- "ASoC: Failed to create bias level debugfs file\n");
+ "ASoC: Failed to create bias level debugfs file: %ld\n",
+ PTR_ERR(d));
}
static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
@@ -2184,10 +2186,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w)
d = debugfs_create_file(w->name, 0444,
dapm->debugfs_dapm, w,
&dapm_widget_power_fops);
- if (!d)
+ if (IS_ERR(d))
dev_warn(w->dapm->dev,
- "ASoC: Failed to create %s debugfs file\n",
- w->name);
+ "ASoC: Failed to create %s debugfs file: %ld\n",
+ w->name, PTR_ERR(d));
}
static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm)
From da7260cc8d1dc3564eb4f33550b0525541d71a47 Mon Sep 17 00:00:00 2001
From: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Date: Wed, 26 Jun 2019 13:49:46 +0300
Subject: [PATCH 13/14] ASoC: codecs: ad193x: Fix memory corruption on BE 64b
systems
Since change_bit() requires unsigned long*, making this cast on an
unsigned int variable will change a wrong bit on BE platforms, causing
memory corruption. Replace this function with a simple XOR.
Fixes: 90f6e6803139 ("ASoC: codecs: ad193x: Fix frame polarity for DSP_A format")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/ad193x.c | 6 ++----
1 file changed, 2 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 96d7cb2e4a56..16e2d334bbe0 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -241,10 +241,8 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/* For DSP_*, LRCLK's polarity must be inverted */
- if (fmt & SND_SOC_DAIFMT_DSP_A) {
- change_bit(ffs(AD193X_DAC_LEFT_HIGH) - 1,
- (unsigned long *)&dac_fmt);
- }
+ if (fmt & SND_SOC_DAIFMT_DSP_A)
+ dac_fmt ^= AD193X_DAC_LEFT_HIGH;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
From 1bcc1fd64e4dd903f4d868a9e053986e3b102713 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Thu, 4 Jul 2019 16:38:50 +0800
Subject: [PATCH 14/14] ASoC: audio-graph-card: fix use-after-free in
graph_for_each_link
After calling of_node_put() on the codec_ep and codec_port variables,
they are still being used, which may result in use-after-free.
We fix this issue by calling of_node_put() after the last usage.
Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2")
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Link: https://lore.kernel.org/r/1562229530-8121-1-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/generic/audio-graph-card.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index ec7e673ba475..70ed28d97d49 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -435,9 +435,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv,
codec_ep = of_graph_get_remote_endpoint(cpu_ep);
codec_port = of_get_parent(codec_ep);
- of_node_put(codec_ep);
- of_node_put(codec_port);
-
/* get convert-xxx property */
memset(&adata, 0, sizeof(adata));
graph_parse_convert(dev, codec_ep, &adata);
@@ -457,6 +454,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv,
else
ret = func_noml(priv, cpu_ep, codec_ep, li);
+ of_node_put(codec_ep);
+ of_node_put(codec_port);
+
if (ret < 0)
return ret;