[PATCH AUTOSEL 4.19 003/172] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[PATCH AUTOSEL 4.19 003/172] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Sasha Levin]

From: Dmitry Osipenko <digetx@gmail.com>

[ Upstream commit 3ef9d5073b552d56bd6daf2af1e89b7e8d4df183 ]

The microphone-jack state needs to be masked in a case of a 4-pin jack
when microphone and ground pins are shorted. Presence of nvidia,headset
tells that WM8903 CODEC driver should mask microphone's status if short
circuit is detected, i.e headphones are inserted.

Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/tegra/tegra_wm8903.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 69bc9461974b..301850df368d 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -173,6 +173,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_component *component = codec_dai->component;
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
+	int shrt = 0;
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det;
@@ -185,12 +186,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 					&tegra_wm8903_hp_jack_gpio);
 	}
 
+	if (of_property_read_bool(card->dev->of_node, "nvidia,headset"))
+		shrt = SND_JACK_MICROPHONE;
+
 	snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
 			      &tegra_wm8903_mic_jack,
 			      tegra_wm8903_mic_jack_pins,
 			      ARRAY_SIZE(tegra_wm8903_mic_jack_pins));
 	wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE,
-				0);
+				shrt);
 
 	snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS");
 
-- 
2.25.1

[PATCH AUTOSEL 4.19 011/172] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type

[Sasha Levin]

From: Xiyu Yang <xiyuyang19@fudan.edu.cn>

[ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ]

davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a
reference of the specified dma_chan object to "chan" with increased
refcnt.

When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes
invalid, so the refcount should be decreased to keep refcount balanced.

The reference counting issue happens in one exception handling path of
davinci_mcasp_get_dma_type(). When chan device is NULL, the function
forgets to decrease the refcnt increased by dma_request_chan(), causing
a refcnt leak.

Fix this issue by calling dma_release_channel() when chan device is
NULL.

Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 14ab16e1369f..1203de8aab99 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1758,8 +1758,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
 				PTR_ERR(chan));
 		return PTR_ERR(chan);
 	}
-	if (WARN_ON(!chan->device || !chan->device->dev))
+	if (WARN_ON(!chan->device || !chan->device->dev)) {
+		dma_release_channel(chan);
 		return -EINVAL;
+	}
 
 	if (chan->device->dev->of_node)
 		ret = of_property_read_string(chan->device->dev->of_node,
-- 
2.25.1

[PATCH AUTOSEL 4.19 014/172] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO

[Sasha Levin]

From: Kai-Heng Feng <kai.heng.feng@canonical.com>

[ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ]

Currently mute LED and micmute LED share the same GPIO polarity.

So split the polarity for mute and micmute, in case they have different
polarities.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/pci/hda/patch_realtek.c | 14 ++++++++------
 1 file changed, 8 insertions(+), 6 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b06f7d52faad..9ab82ead1825 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -94,6 +94,7 @@ struct alc_spec {
 
 	/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
 	int mute_led_polarity;
+	int micmute_led_polarity;
 	hda_nid_t mute_led_nid;
 	hda_nid_t cap_mute_led_nid;
 
@@ -3862,11 +3863,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec,
 
 /* update LED status via GPIO */
 static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
-				bool enabled)
+				int polarity, bool enabled)
 {
-	struct alc_spec *spec = codec->spec;
-
-	if (spec->mute_led_polarity)
+	if (polarity)
 		enabled = !enabled;
 	alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */
 }
@@ -3877,7 +3876,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
 	struct hda_codec *codec = private_data;
 	struct alc_spec *spec = codec->spec;
 
-	alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+	alc_update_gpio_led(codec, spec->gpio_mute_led_mask,
+			    spec->mute_led_polarity, enabled);
 }
 
 /* turn on/off mic-mute LED via GPIO per capture hook */
@@ -3886,6 +3886,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
+			    spec->micmute_led_polarity,
 			    spec->gen.micmute_led.led_value);
 }
 
@@ -5476,7 +5477,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec,
 
 	snd_hda_gen_hp_automute(codec, jack);
 	/* mute_led_polarity is set to 0, so we pass inverted value here */
-	alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present);
+	alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity,
+			    !spec->gen.hp_jack_present);
 }
 
 /* Manage GPIOs for HP EliteBook Folio 9480m.
-- 
2.25.1

[PATCH AUTOSEL 4.19 015/172] ALSA: isa/wavefront: prevent out of bounds write in ioctl

[Sasha Levin]

From: Dan Carpenter <dan.carpenter@oracle.com>

[ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ]

The "header->number" comes from the ioctl and it needs to be clamped to
prevent out of bounds writes.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/isa/wavefront/wavefront_synth.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 0b1e4b34b299..13c8e6542a2f 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header)
 				      "alias for %d\n",
 				      header->number,
 				      header->hdr.a.OriginalSample);
-    
+
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	munge_int32 (header->number, &alias_hdr[0], 2);
 	munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2);
 	munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset),
@@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header)
 	int num_samples;
 	unsigned char *msample_hdr;
 
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL);
 	if (! msample_hdr)
 		return -ENOMEM;
-- 
2.25.1

[PATCH AUTOSEL 4.19 047/172] ASoC: meson: add missing free_irq() in error path

[Sasha Levin]

From: "Pavel Machek (CIP)" <pavel@denx.de>

[ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ]

free_irq() is missing in case of error, fix that.

Signed-off-by: Pavel Machek (CIP) <pavel@denx.de>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>

Link: https://lore.kernel.org/r/20200606153103.GA17905@amd
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/meson/axg-fifo.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 0e4f65e654c4..b229c182b7c3 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -209,7 +209,7 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
 	if (ret)
-		return ret;
+		goto free_irq;
 
 	/* Setup status2 so it reports the memory pointer */
 	regmap_update_bits(fifo->map, FIFO_CTRL1,
@@ -229,8 +229,14 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 	/* Take memory arbitror out of reset */
 	ret = reset_control_deassert(fifo->arb);
 	if (ret)
-		clk_disable_unprepare(fifo->pclk);
+		goto free_clk;
+
+	return 0;
 
+free_clk:
+	clk_disable_unprepare(fifo->pclk);
+free_irq:
+	free_irq(fifo->irq, ss);
 	return ret;
 }
 
-- 
2.25.1

Re: [PATCH AUTOSEL 4.19 047/172] ASoC: meson: add missing free_irq() in error path

[Pavel Machek]

[-- Attachment #1: Type: text/plain, Size: 801 bytes --]

Hi!

> From: "Pavel Machek (CIP)" <pavel@denx.de>
> 
> [ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ]
> 
> free_irq() is missing in case of error, fix that.
> 
> Signed-off-by: Pavel Machek (CIP) <pavel@denx.de>
> Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
> 
> Link: https://lore.kernel.org/r/20200606153103.GA17905@amd
> Signed-off-by: Mark Brown <broonie@kernel.org>

Notice that the bug this fixes is theoretical (found by code review)
and the fix itself was not tested by me as I don't have the hardware.

It may be good idea to for mainline to test the change a bit...

Best regards,
							Pavel
							
-- 
DENX Software Engineering GmbH,      Managing Director: Wolfgang Denk
HRB 165235 Munich, Office: Kirchenstr.5, D-82194 Groebenzell, Germany

[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 181 bytes --]

[PATCH AUTOSEL 4.19 064/172] ALSA: usb-audio: Improve frames size computation

[Sasha Levin]

From: Alexander Tsoy <alexander@tsoy.me>

[ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ]

For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.

But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).

This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.

Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/card.h     |  4 ++++
 sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++-----
 sound/usb/endpoint.h |  1 +
 sound/usb/pcm.c      |  2 ++
 4 files changed, 45 insertions(+), 5 deletions(-)

diff --git a/sound/usb/card.h b/sound/usb/card.h
index 7f11655bde50..36198ddb4754 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -83,6 +83,10 @@ struct snd_usb_endpoint {
 	dma_addr_t sync_dma;		/* DMA address of syncbuf */
 
 	unsigned int pipe;		/* the data i/o pipe */
+	unsigned int framesize[2];	/* small/large frame sizes in samples */
+	unsigned int sample_rem;	/* remainder from division fs/fps */
+	unsigned int sample_accum;	/* sample accumulator */
+	unsigned int fps;		/* frames per second */
 	unsigned int freqn;		/* nominal sampling rate in fs/fps in Q16.16 format */
 	unsigned int freqm;		/* momentary sampling rate in fs/fps in Q16.16 format */
 	int	   freqshift;		/* how much to shift the feedback value to get Q16.16 */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index aeb74cc6ceff..36e255d88937 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep)
 
 /*
  * For streaming based on information derived from sync endpoints,
- * prepare_outbound_urb_sizes() will call next_packet_size() to
+ * prepare_outbound_urb_sizes() will call slave_next_packet_size() to
  * determine the number of samples to be sent in the next packet.
  *
- * For implicit feedback, next_packet_size() is unused.
+ * For implicit feedback, slave_next_packet_size() is unused.
  */
-int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep)
 {
 	unsigned long flags;
 	int ret;
@@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
 	return ret;
 }
 
+/*
+ * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes()
+ * will call next_packet_size() to determine the number of samples to be
+ * sent in the next packet.
+ */
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
+{
+	int ret;
+
+	if (ep->fill_max)
+		return ep->maxframesize;
+
+	ep->sample_accum += ep->sample_rem;
+	if (ep->sample_accum >= ep->fps) {
+		ep->sample_accum -= ep->fps;
+		ret = ep->framesize[1];
+	} else {
+		ret = ep->framesize[0];
+	}
+
+	return ret;
+}
+
 static void retire_outbound_urb(struct snd_usb_endpoint *ep,
 				struct snd_urb_ctx *urb_ctx)
 {
@@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep,
 
 		if (ctx->packet_size[i])
 			counts = ctx->packet_size[i];
+		else if (ep->sync_master)
+			counts = snd_usb_endpoint_slave_next_packet_size(ep);
 		else
 			counts = snd_usb_endpoint_next_packet_size(ep);
 
@@ -887,10 +912,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 	ep->maxpacksize = fmt->maxpacksize;
 	ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX);
 
-	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL)
+	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
 		ep->freqn = get_usb_full_speed_rate(rate);
-	else
+		ep->fps = 1000;
+	} else {
 		ep->freqn = get_usb_high_speed_rate(rate);
+		ep->fps = 8000;
+	}
+
+	ep->sample_rem = rate % ep->fps;
+	ep->framesize[0] = rate / ep->fps;
+	ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
 
 	/* calculate the frequency in 16.16 format */
 	ep->freqm = ep->freqn;
@@ -949,6 +981,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
 	ep->phase = 0;
+	ep->sample_accum = 0;
 
 	snd_usb_endpoint_start_quirk(ep);
 
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 63a39d4fa8d8..d23fa0a8c11b 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_free(struct snd_usb_endpoint *ep);
 
 int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep);
 int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
 
 void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 8b91be394407..42ae774f6db7 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1558,6 +1558,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 	for (i = 0; i < ctx->packets; i++) {
 		if (ctx->packet_size[i])
 			counts = ctx->packet_size[i];
+		else if (ep->sync_master)
+			counts = snd_usb_endpoint_slave_next_packet_size(ep);
 		else
 			counts = snd_usb_endpoint_next_packet_size(ep);
 
-- 
2.25.1

[PATCH AUTOSEL 4.19 065/172] ALSA: usb-audio: Fix racy list management in output queue

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ]

The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.

Do delete the link at the right place inside the spinlock.

Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model")
Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/endpoint.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 36e255d88937..48611849b79b 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -359,17 +359,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
 			ep->next_packet_read_pos %= MAX_URBS;
 
 			/* take URB out of FIFO */
-			if (!list_empty(&ep->ready_playback_urbs))
+			if (!list_empty(&ep->ready_playback_urbs)) {
 				ctx = list_first_entry(&ep->ready_playback_urbs,
 					       struct snd_urb_ctx, ready_list);
+				list_del_init(&ctx->ready_list);
+			}
 		}
 		spin_unlock_irqrestore(&ep->lock, flags);
 
 		if (ctx == NULL)
 			return;
 
-		list_del_init(&ctx->ready_list);
-
 		/* copy over the length information */
 		for (i = 0; i < packet->packets; i++)
 			ctx->packet_size[i] = packet->packet_size[i];
-- 
2.25.1

[PATCH AUTOSEL 4.19 069/172] slimbus: ngd: get drvdata from correct device

[Sasha Levin]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

[ Upstream commit b58c663059b484f7ff547d076a34cf6d7a302e56 ]

Get drvdata directly from parent instead of ngd dev, as ngd
dev can probe defer and previously set drvdata will become null.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200417093618.7929-1-srinivas.kandagatla@linaro.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 drivers/slimbus/qcom-ngd-ctrl.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c
index 9221ba7b7863..f40ac8dcb081 100644
--- a/drivers/slimbus/qcom-ngd-ctrl.c
+++ b/drivers/slimbus/qcom-ngd-ctrl.c
@@ -1350,7 +1350,6 @@ static int of_qcom_slim_ngd_register(struct device *parent,
 		ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME;
 		ngd->pdev->dev.of_node = node;
 		ctrl->ngd = ngd;
-		platform_set_drvdata(ngd->pdev, ctrl);
 
 		platform_device_add(ngd->pdev);
 		ngd->base = ctrl->base + ngd->id * data->offset +
@@ -1365,12 +1364,13 @@ static int of_qcom_slim_ngd_register(struct device *parent,
 
 static int qcom_slim_ngd_probe(struct platform_device *pdev)
 {
-	struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev);
 	struct device *dev = &pdev->dev;
+	struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent);
 	int ret;
 
 	ctrl->ctrl.dev = dev;
 
+	platform_set_drvdata(pdev, ctrl);
 	pm_runtime_use_autosuspend(dev);
 	pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND);
 	pm_runtime_set_suspended(dev);
-- 
2.25.1

[PATCH AUTOSEL 4.19 071/172] soundwire: slave: don't init debugfs on device registration error

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

[ Upstream commit 8893ab5e8ee5d7c12e0fc1dca4a309475064473d ]

The error handling flow seems incorrect, there is no reason to try and
add debugfs support if the device registration did not
succeed. Return on error.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200419185117.4233-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 drivers/soundwire/slave.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c
index ac103bd0c176..b6330b6672d5 100644
--- a/drivers/soundwire/slave.c
+++ b/drivers/soundwire/slave.c
@@ -55,6 +55,8 @@ static int sdw_slave_add(struct sdw_bus *bus,
 		list_del(&slave->node);
 		mutex_unlock(&bus->bus_lock);
 		put_device(&slave->dev);
+
+		return ret;
 	}
 
 	return ret;
-- 
2.25.1

[PATCH AUTOSEL 4.19 098/172] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet

[Sasha Levin]

From: Hans de Goede <hdegoede@redhat.com>

[ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ]

The Toshiba Encore WT8-A tablet almost fully works with the default
settings for non-CR Bay Trail devices. The only problem is that its
jack-detect switch is not inverted (it is active high instead of
the normal active low).

Add a quirk for this model using the default settings +
BYT_RT5640_JD_NOT_INV.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200518072416.5348-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index f29014a7d672..5dfc17987b2b 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -712,6 +712,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Toshiba Encore WT8-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD2_IN4N |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Catch-all for generic Insyde tablets, must be last */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
-- 
2.25.1

[PATCH AUTOSEL 4.19 119/172] ASoC: fix incomplete error-handling in img_i2s_in_probe.

[Sasha Levin]

From: Qiushi Wu <wu000273@umn.edu>

[ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ]

Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()"
if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding
"pm_runtime_put()" into this error path.

Fixes: f65bb92ca12e ("ASoC: img-i2s-in: Add runtime PM")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/img/img-i2s-in.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index 388cefd7340a..c22880aea82a 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -485,6 +485,7 @@ static int img_i2s_in_probe(struct platform_device *pdev)
 	if (IS_ERR(rst)) {
 		if (PTR_ERR(rst) == -EPROBE_DEFER) {
 			ret = -EPROBE_DEFER;
+			pm_runtime_put(&pdev->dev);
 			goto err_suspend;
 		}
 
-- 
2.25.1

[PATCH AUTOSEL 4.19 133/172] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed

[Sasha Levin]

From: Xiyu Yang <xiyuyang19@fudan.edu.cn>

[ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ]

fsl_asrc_dma_hw_params() invokes dma_request_channel() or
fsl_asrc_get_dma_channel(), which returns a reference of the specified
dma_chan object to "pair->dma_chan[dir]" with increased refcnt.

The reference counting issue happens in one exception handling path of
fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End,
the function forgets to decrease the refcnt increased by
dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt
leak.

Fix this issue by calling dma_release_channel() when config DMA channel
failed.

Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/fsl/fsl_asrc_dma.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 1033ac6631b0..b9ac448989ed 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -241,6 +241,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
 	ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
 	if (ret) {
 		dev_err(dev, "failed to config DMA channel for Back-End\n");
+		dma_release_channel(pair->dma_chan[dir]);
 		return ret;
 	}
 
-- 
2.25.1

[PATCH AUTOSEL 4.19 164/172] ASoC: core: only convert non DPCM link to DPCM link

[Sasha Levin]

From: Bard Liao <yung-chuan.liao@linux.intel.com>

[ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ]

Additional checks for valid DAIs expose a corner case, where existing
BE dailinks get modified, e.g. HDMI links are tagged with
dpcm_capture=1 even if the DAIs are for playback.

This patch makes those changes conditional and flags configuration
issues when a BE dailink is has no_pcm=0 but dpcm_playback or
dpcm_capture=1 (which makes no sense).

As discussed on the alsa-devel mailing list, there are redundant flags
for dpcm_playback, dpcm_capture, playback_only, capture_only. This
will have to be cleaned-up in a future update. For now only correct
and flag problematic configurations.

Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture")
Suggested-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-core.c | 22 +++++++++++++++++++---
 1 file changed, 19 insertions(+), 3 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e45dfcb62b6a..595fe20bbc6d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1920,9 +1920,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			dai_link->platform_name = component->name;
 
 			/* convert non BE into BE */
-			dai_link->no_pcm = 1;
-			dai_link->dpcm_playback = 1;
-			dai_link->dpcm_capture = 1;
+			if (!dai_link->no_pcm) {
+				dai_link->no_pcm = 1;
+
+				if (dai_link->dpcm_playback)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
+						 dai_link->name);
+				if (dai_link->dpcm_capture)
+					dev_warn(card->dev,
+						 "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
+						 dai_link->name);
+
+				/* convert normal link into DPCM one */
+				if (!(dai_link->dpcm_playback ||
+				      dai_link->dpcm_capture)) {
+					dai_link->dpcm_playback = !dai_link->capture_only;
+					dai_link->dpcm_capture = !dai_link->playback_only;
+				}
+			}
 
 			/* override any BE fixups */
 			dai_link->be_hw_params_fixup =
-- 
2.25.1

[PATCH AUTOSEL 4.19 165/172] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet

[Sasha Levin]

From: Hans de Goede <hdegoede@redhat.com>

[ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ]

The Toshiba Encore WT10-A tablet almost fully works with the default
settings for Bay Trail CR devices. The only issue is that it uses a
digital mic. connected the the DMIC1 input instead of an analog mic.

Add a quirk for this model using the default settings with the input-map
replaced with BYT_RT5640_DMIC1_MAP.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 5dfc17987b2b..0dcd249877c5 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -724,6 +724,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_JD_NOT_INV |
 					BYT_RT5640_MCLK_EN),
 	},
+	{	/* Toshiba Encore WT10-A */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"),
+		},
+		.driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+					BYT_RT5640_JD_SRC_JD1_IN4P |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_0P75 |
+					BYT_RT5640_SSP0_AIF2 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{	/* Catch-all for generic Insyde tablets, must be last */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
-- 
2.25.1

[PATCH AUTOSEL 4.19 166/172] ASoC: rt5645: Add platform-data for Asus T101HA

[Sasha Levin]

From: Hans de Goede <hdegoede@redhat.com>

[ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ]

The Asus T101HA uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like the Asus T100HA. Note unlike the T100HA its jack-detect is not
inverted.

Add a DMI quirk with the correct settings for this model.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/rt5645.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 7e3b47eeea04..9185bd7c5a6d 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3656,6 +3656,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = {
 	.inv_jd1_1 = true,
 };
 
+static const struct rt5645_platform_data asus_t101ha_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+	.dmic2_data_pin = RT5645_DMIC2_DISABLE,
+	.jd_mode = 3,
+};
+
 static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = {
 	.jd_mode = 3,
 	.in2_diff = true,
@@ -3733,6 +3739,14 @@ static const struct dmi_system_id dmi_platform_data[] = {
 		},
 		.driver_data = (void *)&asus_t100ha_platform_data,
 	},
+	{
+		.ident = "ASUS T101HA",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"),
+		},
+		.driver_data = (void *)&asus_t101ha_platform_data,
+	},
 	{
 		.ident = "MINIX Z83-4",
 		.matches = {
-- 
2.25.1