[PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[PATCH AUTOSEL 5.7 004/388] 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 9b5651502f12..3aca354f9e08 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -177,6 +177,7 @@ 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;
@@ -189,12 +190,15 @@ 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 5.7 007/388] ASoC: SOF: imx8: Fix randbuild error

[Sasha Levin]

From: YueHaibing <yuehaibing@huawei.com>

[ Upstream commit fe17e6cdc0fefca96ba9659be4b2b07487cbf0c5 ]

when do randconfig like this:
CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y
CONFIG_SND_SOC_SOF_IMX8=y
CONFIG_SND_SOC_SOF_OF=y
CONFIG_IMX_DSP=m
CONFIG_IMX_SCU=y

there is a link error:

sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg':
imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell'

Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this

Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/imx/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig
index bae4f7bf5f75..812749064ca8 100644
--- a/sound/soc/sof/imx/Kconfig
+++ b/sound/soc/sof/imx/Kconfig
@@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL
 config SND_SOC_SOF_IMX8_SUPPORT
 	bool "SOF support for i.MX8"
 	depends on IMX_SCU
-	depends on IMX_DSP
+	select IMX_DSP
 	help
 	  This adds support for Sound Open Firmware for NXP i.MX8 platforms
 	  Say Y if you have such a device.
-- 
2.25.1

[PATCH AUTOSEL 5.7 013/388] ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet

[Sasha Levin]

From: Shengjiu Wang <shengjiu.wang@nxp.com>

[ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ]

Disable exception interrupt before scheduling tasklet, otherwise if
the tasklet isn't handled immediately, there will be endless xrun
interrupt.

Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.1587968824.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/fsl/fsl_esai.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c7a49d03463a..84290be778f0 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -87,6 +87,10 @@ static irqreturn_t esai_isr(int irq, void *devid)
 	if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) &&
 	    esai_priv->reset_at_xrun) {
 		dev_dbg(&pdev->dev, "reset module for xrun\n");
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR,
+				   ESAI_xCR_xEIE_MASK, 0);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR,
+				   ESAI_xCR_xEIE_MASK, 0);
 		tasklet_schedule(&esai_priv->task);
 	}
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 016/388] 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/ti/davinci-mcasp.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 734ffe925c4d..7a7db743dc5b 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1896,8 +1896,10 @@ 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 5.7 020/388] ASoC: codecs: wm97xx: fix ac97 dependency

[Sasha Levin]

From: Arnd Bergmann <arnd@arndb.de>

[ Upstream commit ee2cbe06935bfa58f1fe07dc2a2283945f4b97dc ]

A recent build fix got the dependency slightly wrong, breaking
builds with CONFIG_AC97_BUS_NEW:

WARNING: unmet direct dependencies detected for SND_SOC_WM9713
  Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n]
  Selected by [m]:
  - SND_SOC_ZYLONITE [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && MACH_ZYLONITE [=y] && AC97_BUS [=n]=n

WARNING: unmet direct dependencies detected for SND_SOC_WM9712
  Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n]
  Selected by [m]:
  - SND_PXA2XX_SOC_EM_X270 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && (MACH_EM_X270 [=n] || MACH_EXEDA [=n] || MACH_CM_X300 [=y]) && AC97_BUS [=n]=n

Change the dependency to allow either version of the AC97 library
code.

Fixes: 5a309875787d ("ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200428212721.2877627-1-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/Kconfig | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e60e0b6a689c..8cdc68c141dc 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1620,19 +1620,19 @@ config SND_SOC_WM9090
 
 config SND_SOC_WM9705
 	tristate
-	depends on SND_SOC_AC97_BUS
+	depends on SND_SOC_AC97_BUS || AC97_BUS_NEW
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
 config SND_SOC_WM9712
 	tristate
-	depends on SND_SOC_AC97_BUS
+	depends on SND_SOC_AC97_BUS || AC97_BUS_NEW
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
 config SND_SOC_WM9713
 	tristate
-	depends on SND_SOC_AC97_BUS
+	depends on SND_SOC_AC97_BUS || AC97_BUS_NEW
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 025/388] 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 2c4575909441..e057ecb5a904 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -81,6 +81,7 @@ 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;
 
@@ -4080,11 +4081,9 @@ 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 */
 }
@@ -4095,7 +4094,8 @@ 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 */
@@ -4104,6 +4104,7 @@ 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);
 }
 
@@ -5808,7 +5809,8 @@ 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 5.7 026/388] 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 c5b1d5900eed..d6420d224d09 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1171,7 +1171,10 @@ 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),
@@ -1202,6 +1205,9 @@ 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 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set

[Sasha Levin]

From: Daniel Baluta <daniel.baluta@nxp.com>

[ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ]

This provides a better separation between runtime and PM sleep
callbacks.

Only do nothing if given runtime flag is set and calback is not set.

With the current implementation, if PM sleep callback is set but runtime
callback is not set then at runtime resume we reload the firmware even
if we do not support runtime resume callback.

Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200515135958.17511-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/pm.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index c410822d9920..01d83ddc16ba 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -90,7 +90,10 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 	int ret;
 
 	/* do nothing if dsp resume callbacks are not set */
-	if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume)
+	if (!runtime_resume && !sof_ops(sdev)->resume)
+		return 0;
+
+	if (runtime_resume && !sof_ops(sdev)->runtime_resume)
 		return 0;
 
 	/* DSP was never successfully started, nothing to resume */
@@ -175,7 +178,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	int ret;
 
 	/* do nothing if dsp suspend callback is not set */
-	if (!sof_ops(sdev)->suspend)
+	if (!runtime_suspend && !sof_ops(sdev)->suspend)
+		return 0;
+
+	if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
 		return 0;
 
 	if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
-- 
2.25.1

[PATCH AUTOSEL 5.7 085/388] ASoC: qcom: q6asm-dai: kCFI fix

[Sasha Levin]

From: John Stultz <john.stultz@linaro.org>

[ Upstream commit a6b675a89e51a1cdad0481b809b7840d3f86e4b5 ]

Fixes the following kCFI crash seen on db845c, caused
by the function prototypes not matching the callback
function prototype.

[   82.585661] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000001
[   82.595387] Mem abort info:
[   82.599463]   ESR = 0x96000005
[   82.602658]   EC = 0x25: DABT (current EL), IL = 32 bits
[   82.608177]   SET = 0, FnV = 0
[   82.611829]   EA = 0, S1PTW = 0
[   82.615369] Data abort info:
[   82.618751]   ISV = 0, ISS = 0x00000005
[   82.622641]   CM = 0, WnR = 0
[   82.625774] user pgtable: 4k pages, 39-bit VAs, pgdp=0000000174259000
[   82.632292] [0000000000000001] pgd=0000000000000000, pud=0000000000000000
[   82.639167] Internal error: Oops: 96000005 [#1] PREEMPT SMP
[   82.644795] Modules linked in: hci_uart btqca xhci_plat_hcd xhci_pci_renesas xhci_pci xhci_hcd wcn36xx wcnss_ctrl wcd934x vctrl_regulator ufs_qcom syscon_reboot_e
[   82.644927]  qcom_apcs_ipc_mailbox q6asm_dai q6routing q6asm q6afe_dai q6adm q6afe q6core q6dsp_common pm8941_pwrkey pm8916_wdt platform_mhu pinctrl_spmi_mpp pine
[   82.812982] CPU: 3 PID: 240 Comm: kworker/u16:4 Tainted: G        W         5.6.0-rc7-mainline-00960-g0c34353d11b9-dirty #1
[   82.824201] Hardware name: Thundercomm Dragonboard 845c (DT)
[   82.829937] Workqueue: qcom_apr_rx apr_rxwq [apr]
[   82.834698] pstate: 80c00005 (Nzcv daif +PAN +UAO)
[   82.839553] pc : __cfi_check_fail+0x4/0x1c [q6asm_dai]
[   82.844754] lr : __cfi_check+0x3a8/0x3b0 [q6asm_dai]
[   82.849767] sp : ffffffc0105f3c20
[   82.853123] x29: ffffffc0105f3c30 x28: 0000000000000020
[   82.858489] x27: ffffff80f4588400 x26: ffffff80f458ec94
[   82.863854] x25: ffffff80f458ece8 x24: ffffffe3670c7000
[   82.869220] x23: ffffff8094bb7b34 x22: ffffffe367137000
[   82.874585] x21: bd07909b332eada6 x20: 0000000000000001
[   82.879950] x19: ffffffe36713863c x18: ffffff80f8df4430
[   82.885316] x17: 0000000000000001 x16: ffffffe39d15e660
[   82.890681] x15: 0000000000000001 x14: 0000000000000027
[   82.896047] x13: 0000000000000000 x12: ffffffe39e6465a0
[   82.901413] x11: 0000000000000051 x10: 000000000000ffff
[   82.906779] x9 : 000ffffffe366c19 x8 : c3c5f18762d1ceef
[   82.912145] x7 : 0000000000000000 x6 : ffffffc010877698
[   82.917511] x5 : ffffffc0105f3c00 x4 : 0000000000000000
[   82.922877] x3 : 0000000000000000 x2 : 0000000000000001
[   82.928243] x1 : ffffffe36713863c x0 : 0000000000000001
[   82.933610] Call trace:
[   82.936099]  __cfi_check_fail+0x4/0x1c [q6asm_dai]
[   82.940955]  q6asm_srvc_callback+0x22c/0x618 [q6asm]
[   82.945973]  apr_rxwq+0x1a8/0x27c [apr]
[   82.949861]  process_one_work+0x2e8/0x54c
[   82.953919]  worker_thread+0x27c/0x4d4
[   82.957715]  kthread+0x144/0x154
[   82.960985]  ret_from_fork+0x10/0x18
[   82.964603] Code: a8c37bfd f85f8e5e d65f03c0 b40000a0 (39400008)
[   82.970762] ---[ end trace 410accb839617143 ]---
[   82.975429] Kernel panic - not syncing: Fatal exception

Signed-off-by: John Stultz <john.stultz@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
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: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Stephan Gerhold <stephan@gerhold.net>
Cc: Sami Tolvanen <samitolvanen@google.com>
Cc: Todd Kjos <tkjos@google.com>
Cc: Alistair Delva <adelva@google.com>
Cc: Amit Pundir <amit.pundir@linaro.org>
Cc: Sumit Semwal <sumit.semwal@linaro.org>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200529213823.98812-1-john.stultz@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 125af00bba53..4640804aab7f 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -176,7 +176,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = {
 };
 
 static void event_handler(uint32_t opcode, uint32_t token,
-			  uint32_t *payload, void *priv)
+			  void *payload, void *priv)
 {
 	struct q6asm_dai_rtd *prtd = priv;
 	struct snd_pcm_substream *substream = prtd->substream;
@@ -490,7 +490,7 @@ static int q6asm_dai_hw_params(struct snd_soc_component *component,
 }
 
 static void compress_event_handler(uint32_t opcode, uint32_t token,
-				   uint32_t *payload, void *priv)
+				   void *payload, void *priv)
 {
 	struct q6asm_dai_rtd *prtd = priv;
 	struct snd_compr_stream *substream = prtd->cstream;
-- 
2.25.1

[PATCH AUTOSEL 5.7 094/388] ASoC: meson: fix memory leak of links if allocation of ldata fails

[Sasha Levin]

From: Colin Ian King <colin.king@canonical.com>

[ Upstream commit 6e801dc411329aff592fbd48fb116183d0acdb00 ]

Currently if the allocation of ldata fails the error return path
does not kfree the allocated links object.  Fix this by adding
an error exit return path that performs the necessary kfree'ing.

Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Addresses-Coverity: ("Resource leak")
Link: https://lore.kernel.org/r/20200604171216.60043-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/meson/meson-card-utils.c | 17 ++++++++++++-----
 1 file changed, 12 insertions(+), 5 deletions(-)

diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index 2ca8c98e204f..5a4a91c88734 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card,
 	links = krealloc(priv->card.dai_link,
 			 num_links * sizeof(*priv->card.dai_link),
 			 GFP_KERNEL | __GFP_ZERO);
+	if (!links)
+		goto err_links;
+
 	ldata = krealloc(priv->link_data,
 			 num_links * sizeof(*priv->link_data),
 			 GFP_KERNEL | __GFP_ZERO);
-
-	if (!links || !ldata) {
-		dev_err(priv->card.dev, "failed to allocate links\n");
-		return -ENOMEM;
-	}
+	if (!ldata)
+		goto err_ldata;
 
 	priv->card.dai_link = links;
 	priv->link_data = ldata;
 	priv->card.num_links = num_links;
 	return 0;
+
+err_ldata:
+	kfree(links);
+err_links:
+	dev_err(priv->card.dev, "failed to allocate links\n");
+	return -ENOMEM;
+
 }
 EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 103/388] 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 2e9b56b29d31..b2e867113226 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* 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,
@@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component,
 	/* 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;
 }
 EXPORT_SYMBOL_GPL(axg_fifo_pcm_open);
-- 
2.25.1

[PATCH AUTOSEL 5.7 124/388] ALSA: usb-audio: RME Babyface Pro mixer patch

[Sasha Levin]

From: Thomas Ebeling <penguins@bollie.de>

[ Upstream commit 3e8f3bd047163d30fb1ad32ca7e4628921555c09 ]

Added mixer quirks to allow controlling the internal DSP of the
RME Babyface Pro and its successor Babyface Pro FS.

Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/mixer_quirks.c | 418 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 418 insertions(+)

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index a5f65a9a0254..bdff8674942e 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2185,6 +2185,421 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+/*
+ * RME Babyface Pro (FS)
+ *
+ * These devices exposes a couple of DSP functions via request to EP0.
+ * Switches are available via control registers, while routing is controlled
+ * by controlling the volume on each possible crossing point.
+ * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with
+ * 0dB being at dec. 32768.
+ */
+enum {
+	SND_BBFPRO_CTL_REG1 = 0,
+	SND_BBFPRO_CTL_REG2
+};
+
+#define SND_BBFPRO_CTL_REG_MASK 1
+#define SND_BBFPRO_CTL_IDX_MASK 0xff
+#define SND_BBFPRO_CTL_IDX_SHIFT 1
+#define SND_BBFPRO_CTL_VAL_MASK 1
+#define SND_BBFPRO_CTL_VAL_SHIFT 9
+#define SND_BBFPRO_CTL_REG1_CLK_MASTER 0
+#define SND_BBFPRO_CTL_REG1_CLK_OPTICAL 1
+#define SND_BBFPRO_CTL_REG1_SPDIF_PRO 7
+#define SND_BBFPRO_CTL_REG1_SPDIF_EMPH 8
+#define SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL 10
+#define SND_BBFPRO_CTL_REG2_48V_AN1 0
+#define SND_BBFPRO_CTL_REG2_48V_AN2 1
+#define SND_BBFPRO_CTL_REG2_SENS_IN3 2
+#define SND_BBFPRO_CTL_REG2_SENS_IN4 3
+#define SND_BBFPRO_CTL_REG2_PAD_AN1 4
+#define SND_BBFPRO_CTL_REG2_PAD_AN2 5
+
+#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff
+#define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff
+#define SND_BBFPRO_MIXER_VAL_SHIFT 9
+#define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf
+#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB
+
+#define SND_BBFPRO_USBREQ_CTL_REG1 0x10
+#define SND_BBFPRO_USBREQ_CTL_REG2 0x17
+#define SND_BBFPRO_USBREQ_MIXER 0x12
+
+static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg,
+				 u8 index, u8 value)
+{
+	int err;
+	u16 usb_req, usb_idx, usb_val;
+	struct snd_usb_audio *chip = mixer->chip;
+
+	err = snd_usb_lock_shutdown(chip);
+	if (err < 0)
+		return err;
+
+	if (reg == SND_BBFPRO_CTL_REG1) {
+		usb_req = SND_BBFPRO_USBREQ_CTL_REG1;
+		if (index == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) {
+			usb_idx = 3;
+			usb_val = value ? 3 : 0;
+		} else {
+			usb_idx = 1 << index;
+			usb_val = value ? usb_idx : 0;
+		}
+	} else {
+		usb_req = SND_BBFPRO_USBREQ_CTL_REG2;
+		usb_idx = 1 << index;
+		usb_val = value ? usb_idx : 0;
+	}
+
+	err = snd_usb_ctl_msg(chip->dev,
+			      usb_sndctrlpipe(chip->dev, 0), usb_req,
+			      USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+			      usb_val, usb_idx, 0, 0);
+
+	snd_usb_unlock_shutdown(chip);
+	return err;
+}
+
+static int snd_bbfpro_ctl_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	u8 reg, idx, val;
+	int pv;
+
+	pv = kcontrol->private_value;
+	reg = pv & SND_BBFPRO_CTL_REG_MASK;
+	idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK;
+	val = kcontrol->private_value >> SND_BBFPRO_CTL_VAL_SHIFT;
+
+	if ((reg == SND_BBFPRO_CTL_REG1 &&
+	     idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) ||
+	    (reg == SND_BBFPRO_CTL_REG2 &&
+	    (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 ||
+	     idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) {
+		ucontrol->value.enumerated.item[0] = val;
+	} else {
+		ucontrol->value.integer.value[0] = val;
+	}
+	return 0;
+}
+
+static int snd_bbfpro_ctl_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	u8 reg, idx;
+	int pv;
+
+	pv = kcontrol->private_value;
+	reg = pv & SND_BBFPRO_CTL_REG_MASK;
+	idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK;
+
+	if (reg == SND_BBFPRO_CTL_REG1 &&
+	    idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) {
+		static const char * const texts[2] = {
+			"AutoSync",
+			"Internal"
+		};
+		return snd_ctl_enum_info(uinfo, 1, 2, texts);
+	} else if (reg == SND_BBFPRO_CTL_REG2 &&
+		   (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 ||
+		    idx == SND_BBFPRO_CTL_REG2_SENS_IN4)) {
+		static const char * const texts[2] = {
+			"-10dBV",
+			"+4dBu"
+		};
+		return snd_ctl_enum_info(uinfo, 1, 2, texts);
+	}
+
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	return 0;
+}
+
+static int snd_bbfpro_ctl_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	int err;
+	u8 reg, idx;
+	int old_value, pv, val;
+
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+	struct usb_mixer_interface *mixer = list->mixer;
+
+	pv = kcontrol->private_value;
+	reg = pv & SND_BBFPRO_CTL_REG_MASK;
+	idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK;
+	old_value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK;
+
+	if ((reg == SND_BBFPRO_CTL_REG1 &&
+	     idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) ||
+	    (reg == SND_BBFPRO_CTL_REG2 &&
+	    (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 ||
+	     idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) {
+		val = ucontrol->value.enumerated.item[0];
+	} else {
+		val = ucontrol->value.integer.value[0];
+	}
+
+	if (val > 1)
+		return -EINVAL;
+
+	if (val == old_value)
+		return 0;
+
+	kcontrol->private_value = reg
+		| ((idx & SND_BBFPRO_CTL_IDX_MASK) << SND_BBFPRO_CTL_IDX_SHIFT)
+		| ((val & SND_BBFPRO_CTL_VAL_MASK) << SND_BBFPRO_CTL_VAL_SHIFT);
+
+	err = snd_bbfpro_ctl_update(mixer, reg, idx, val);
+	return err < 0 ? err : 1;
+}
+
+static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list)
+{
+	u8 reg, idx;
+	int value, pv;
+
+	pv = list->kctl->private_value;
+	reg = pv & SND_BBFPRO_CTL_REG_MASK;
+	idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK;
+	value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK;
+
+	return snd_bbfpro_ctl_update(list->mixer, reg, idx, value);
+}
+
+static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index,
+				 u32 value)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	int err;
+	u16 idx;
+	u16 usb_idx, usb_val;
+	u32 v;
+
+	err = snd_usb_lock_shutdown(chip);
+	if (err < 0)
+		return err;
+
+	idx = index & SND_BBFPRO_MIXER_IDX_MASK;
+	// 18 bit linear volume, split so 2 bits end up in index.
+	v = value & SND_BBFPRO_MIXER_VAL_MASK;
+	usb_idx = idx | (v & 0x3) << 14;
+	usb_val = (v >> 2) & 0xffff;
+
+	err = snd_usb_ctl_msg(chip->dev,
+			      usb_sndctrlpipe(chip->dev, 0),
+			      SND_BBFPRO_USBREQ_MIXER,
+			      USB_DIR_OUT | USB_TYPE_VENDOR |
+			      USB_RECIP_DEVICE,
+			      usb_val, usb_idx, 0, 0);
+
+	snd_usb_unlock_shutdown(chip);
+	return err;
+}
+
+static int snd_bbfpro_vol_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] =
+		kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT;
+	return 0;
+}
+
+static int snd_bbfpro_vol_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = SND_BBFPRO_MIXER_VAL_MIN;
+	uinfo->value.integer.max = SND_BBFPRO_MIXER_VAL_MAX;
+	return 0;
+}
+
+static int snd_bbfpro_vol_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	int err;
+	u16 idx;
+	u32 new_val, old_value, uvalue;
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);
+	struct usb_mixer_interface *mixer = list->mixer;
+
+	uvalue = ucontrol->value.integer.value[0];
+	idx = kcontrol->private_value & SND_BBFPRO_MIXER_IDX_MASK;
+	old_value = kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT;
+
+	if (uvalue > SND_BBFPRO_MIXER_VAL_MAX)
+		return -EINVAL;
+
+	if (uvalue == old_value)
+		return 0;
+
+	new_val = uvalue & SND_BBFPRO_MIXER_VAL_MASK;
+
+	kcontrol->private_value = idx
+		| (new_val << SND_BBFPRO_MIXER_VAL_SHIFT);
+
+	err = snd_bbfpro_vol_update(mixer, idx, new_val);
+	return err < 0 ? err : 1;
+}
+
+static int snd_bbfpro_vol_resume(struct usb_mixer_elem_list *list)
+{
+	int pv = list->kctl->private_value;
+	u16 idx = pv & SND_BBFPRO_MIXER_IDX_MASK;
+	u32 val = (pv >> SND_BBFPRO_MIXER_VAL_SHIFT)
+		& SND_BBFPRO_MIXER_VAL_MASK;
+	return snd_bbfpro_vol_update(list->mixer, idx, val);
+}
+
+// Predfine elements
+static const struct snd_kcontrol_new snd_bbfpro_ctl_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.index = 0,
+	.info = snd_bbfpro_ctl_info,
+	.get = snd_bbfpro_ctl_get,
+	.put = snd_bbfpro_ctl_put
+};
+
+static const struct snd_kcontrol_new snd_bbfpro_vol_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.index = 0,
+	.info = snd_bbfpro_vol_info,
+	.get = snd_bbfpro_vol_get,
+	.put = snd_bbfpro_vol_put
+};
+
+static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg,
+			      u8 index, char *name)
+{
+	struct snd_kcontrol_new knew = snd_bbfpro_ctl_control;
+
+	knew.name = name;
+	knew.private_value = (reg & SND_BBFPRO_CTL_REG_MASK)
+		| ((index & SND_BBFPRO_CTL_IDX_MASK)
+			<< SND_BBFPRO_CTL_IDX_SHIFT);
+
+	return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_ctl_resume,
+		&knew, NULL);
+}
+
+static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index,
+			      char *name)
+{
+	struct snd_kcontrol_new knew = snd_bbfpro_vol_control;
+
+	knew.name = name;
+	knew.private_value = index & SND_BBFPRO_MIXER_IDX_MASK;
+
+	return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_vol_resume,
+		&knew, NULL);
+}
+
+static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer)
+{
+	int err, i, o;
+	char name[48];
+
+	static const char * const input[] = {
+		"AN1", "AN2", "IN3", "IN4", "AS1", "AS2", "ADAT3",
+		"ADAT4", "ADAT5", "ADAT6", "ADAT7", "ADAT8"};
+
+	static const char * const output[] = {
+		"AN1", "AN2", "PH3", "PH4", "AS1", "AS2", "ADAT3", "ADAT4",
+		"ADAT5", "ADAT6", "ADAT7", "ADAT8"};
+
+	for (o = 0 ; o < 12 ; ++o) {
+		for (i = 0 ; i < 12 ; ++i) {
+			// Line routing
+			snprintf(name, sizeof(name),
+				 "%s-%s-%s Playback Volume",
+				 (i < 2 ? "Mic" : "Line"),
+				 input[i], output[o]);
+			err = snd_bbfpro_vol_add(mixer, (26 * o + i), name);
+			if (err < 0)
+				return err;
+
+			// PCM routing... yes, it is output remapping
+			snprintf(name, sizeof(name),
+				 "PCM-%s-%s Playback Volume",
+				 output[i], output[o]);
+			err = snd_bbfpro_vol_add(mixer, (26 * o + 12 + i),
+						 name);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	// Control Reg 1
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1,
+				 SND_BBFPRO_CTL_REG1_CLK_OPTICAL,
+				 "Sample Clock Source");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1,
+				 SND_BBFPRO_CTL_REG1_SPDIF_PRO,
+				 "IEC958 Pro Mask");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1,
+				 SND_BBFPRO_CTL_REG1_SPDIF_EMPH,
+				 "IEC958 Emphasis");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1,
+				 SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL,
+				 "IEC958 Switch");
+	if (err < 0)
+		return err;
+
+	// Control Reg 2
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_48V_AN1,
+				 "Mic-AN1 48V");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_48V_AN2,
+				 "Mic-AN2 48V");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_SENS_IN3,
+				 "Line-IN3 Sens.");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_SENS_IN4,
+				 "Line-IN4 Sens.");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_PAD_AN1,
+				 "Mic-AN1 PAD");
+	if (err < 0)
+		return err;
+
+	err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2,
+				 SND_BBFPRO_CTL_REG2_PAD_AN2,
+				 "Mic-AN2 PAD");
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -2286,6 +2701,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */
 		err = snd_sc1810_init_mixer(mixer);
 		break;
+	case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */
+		err = snd_bbfpro_controls_create(mixer);
+		break;
 	}
 
 	return err;
-- 
2.25.1

[PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983

[Sasha Levin]

From: Wei Li <liwei391@huawei.com>

[ Upstream commit c1c050ee74d67aeb879fd38e3a07139d7fdb79f4 ]

As these two drivers support I2C and SPI, we should add the SND_SOC_I2C_AND_SPI
dependency instead.

Fixes: ce0c97f8a2936 ("ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout")
Signed-off-by: Wei Li <liwei391@huawei.com>
Link: https://lore.kernel.org/r/20200420202410.47327-3-liwei391@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/Kconfig | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8cdc68c141dc..7d2cbed55a9d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -717,7 +717,7 @@ config SND_SOC_L3
 
 config SND_SOC_DA7210
 	tristate
-	depends on I2C
+	depends on SND_SOC_I2C_AND_SPI
 
 config SND_SOC_DA7213
 	tristate "Dialog DA7213 CODEC"
@@ -1569,7 +1569,7 @@ config SND_SOC_WM8978
 
 config SND_SOC_WM8983
 	tristate
-	depends on I2C
+	depends on SND_SOC_I2C_AND_SPI
 
 config SND_SOC_WM8985
 	tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver"
-- 
2.25.1

[PATCH AUTOSEL 5.7 138/388] 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 395403a2d33f..820e564656ed 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -84,6 +84,10 @@ 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 4a9a2f6ef5a4..d8dc7cb56d43 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -124,12 +124,12 @@ 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;
@@ -146,6 +146,29 @@ 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)
 {
@@ -190,6 +213,8 @@ 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);
 
@@ -874,10 +899,17 @@ 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;
@@ -936,6 +968,7 @@ 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 a4e4064f9aee..b50965ab3b3a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1579,6 +1579,8 @@ 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 5.7 139/388] 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 d8dc7cb56d43..50104f658ed4 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -346,17 +346,17 @@ 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 5.7 147/388] 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 fc2575fef51b..7426b5884218 100644
--- a/drivers/slimbus/qcom-ngd-ctrl.c
+++ b/drivers/slimbus/qcom-ngd-ctrl.c
@@ -1361,7 +1361,6 @@ 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 +
@@ -1376,12 +1375,13 @@ 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 5.7 155/388] ASoC: component: suppress uninitialized-variable warning

[Sasha Levin]

From: Arnd Bergmann <arnd@arndb.de>

[ Upstream commit be16a0f0dc8fab8e25d9cdbeb4f8f28afc9186d2 ]

Old versions of gcc (tested on gcc-4.8) produce a warning for
correct code:

sound/soc/soc-compress.c: In function 'soc_compr_open':
sound/soc/soc-compress.c:75:28: error: 'component' is used uninitialized in this function [-Werror=uninitialized]
  struct snd_soc_component *component, *save = NULL;

Change the for_each_rtd_components() macro to ensure 'component'
gets initialized to a value the compiler does not complain about.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200428214754.3925368-1-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 include/sound/soc.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 946f88a6c63d..e0371e70242d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1177,7 +1177,7 @@ struct snd_soc_pcm_runtime {
 #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
 
 #define for_each_rtd_components(rtd, i, component)			\
-	for ((i) = 0;							\
+	for ((i) = 0, component = NULL;					\
 	     ((i) < rtd->num_components) && ((component) = rtd->components[i]);\
 	     (i)++)
 #define for_each_rtd_cpu_dais(rtd, i, dai)				\
-- 
2.25.1

[PATCH AUTOSEL 5.7 157/388] ASoC: rt5682: fix I2C/Soundwire dependencies

[Sasha Levin]

From: Arnd Bergmann <arnd@arndb.de>

[ Upstream commit fd443a20c2f0950f3c31765a08f7dd49b3bc69cb ]

If one of the two is a loadable module, the combined driver must
not be built-in:

aarch64-linux-ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_free':
rt5682.c:(.text+0xb34): undefined reference to `sdw_stream_remove_slave'
aarch64-linux-ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_params':
rt5682.c:(.text+0xe78): undefined reference to `sdw_stream_add_slave'

In particular, the soundwire driver must not be built-in if
CONFIG_I2C=m.

Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200428214642.3925004-1-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/Kconfig | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7d2cbed55a9d..4c643d2e6d6b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1136,10 +1136,13 @@ config SND_SOC_RT5677_SPI
 config SND_SOC_RT5682
 	tristate
 	depends on I2C || SOUNDWIRE
+	depends on SOUNDWIRE || !SOUNDWIRE
+	depends on I2C || !I2C
 
 config SND_SOC_RT5682_SDW
 	tristate "Realtek RT5682 Codec - SDW"
 	depends on SOUNDWIRE
+	depends on I2C || !I2C
 	select SND_SOC_RT5682
 	select REGMAP_SOUNDWIRE
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 160/388] ASoC: SOF: Update correct LED status at the first time usage of update_mute_led()

[Sasha Levin]

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

[ Upstream commit 49c22696348d6e7c8a2ecfd7e60fddfe188ded82 ]

At the first time update_mute_led() gets called, if channels are already
muted, the temp value equals to led_value as 0, skipping the following
LED setting.

So set led_value to -1 as an uninitialized state, to update the correct
LED status at first time usage.

Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200430091139.7003-1-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/control.c   | 4 ++--
 sound/soc/sof/sof-audio.h | 2 +-
 sound/soc/sof/topology.c  | 2 ++
 3 files changed, 5 insertions(+), 3 deletions(-)

diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c
index dfc412e2d956..6d63768d42aa 100644
--- a/sound/soc/sof/control.c
+++ b/sound/soc/sof/control.c
@@ -19,8 +19,8 @@ static void update_mute_led(struct snd_sof_control *scontrol,
 			    struct snd_kcontrol *kcontrol,
 			    struct snd_ctl_elem_value *ucontrol)
 {
-	unsigned int temp = 0;
-	unsigned int mask;
+	int temp = 0;
+	int mask;
 	int i;
 
 	mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index bf65f31af858..875a5fc13297 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -56,7 +56,7 @@ struct snd_sof_pcm {
 struct snd_sof_led_control {
 	unsigned int use_led;
 	unsigned int direction;
-	unsigned int led_value;
+	int led_value;
 };
 
 /* ALSA SOF Kcontrol device */
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index fe8ba3e05e08..ab2b69de1d4d 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1203,6 +1203,8 @@ static int sof_control_load(struct snd_soc_component *scomp, int index,
 		return ret;
 	}
 
+	scontrol->led_ctl.led_value = -1;
+
 	dobj->private = scontrol;
 	list_add(&scontrol->list, &sdev->kcontrol_list);
 	return ret;
-- 
2.25.1

[PATCH AUTOSEL 5.7 162/388] ASoC: max98373: reorder max98373_reset() in resume

[Sasha Levin]

From: Yong Zhi <yong.zhi@intel.com>

[ Upstream commit 1a446873d7dd3a450f685928ce7f1907bde4583d ]

During S3 test, the following error was observed:

[ 726.174237] i2c_designware i2c_designware.0: platform_pm_resume+0x0/0x3d returned 0 after 0 usecs
[ 726.184187] max98373 i2c-MX98373:00: calling max98373_resume+0x0/0x30 [snd_soc_max98373] @ 12698, parent: i2c-11
[ 726.195589] max98373 i2c-MX98373:00: Reset command failed. (ret:-16)

When calling regmap_update_bits(), since map->reg_update_bits is NULL,
_regmap_read() is entered with the following logic:

	if (!map->cache_bypass) {
		ret = regcache_read(map, reg, val);
		if (ret == 0)
			return 0;
	}

	if (map->cache_only)
		return -EBUSY;

regcache_read() hits -EINVAL because MAX98373_R2000_SW_RESET is volatile,
as map->cache_only is set by codec suspend, thus -EBUSY is returned.
Fix by moving max98373_reset() after cache_only set to false in max98373_resume().

Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Link: https://lore.kernel.org/r/1588376661-29799-1-git-send-email-yong.zhi@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/max98373.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index cae1def8902d..96718e3a1ad0 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -850,8 +850,8 @@ static int max98373_resume(struct device *dev)
 {
 	struct max98373_priv *max98373 = dev_get_drvdata(dev);
 
-	max98373_reset(max98373, dev);
 	regcache_cache_only(max98373->regmap, false);
+	max98373_reset(max98373, dev);
 	regcache_sync(max98373->regmap);
 	return 0;
 }
-- 
2.25.1

[PATCH AUTOSEL 5.7 163/388] 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 aace57fae7f8..4bacdb187eab 100644
--- a/drivers/soundwire/slave.c
+++ b/drivers/soundwire/slave.c
@@ -68,6 +68,8 @@ static int sdw_slave_add(struct sdw_bus *bus,
 		list_del(&slave->node);
 		mutex_unlock(&bus->bus_lock);
 		put_device(&slave->dev);
+
+		return ret;
 	}
 	sdw_slave_debugfs_init(slave);
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 179/388] ALSA: firewire-lib: fix invalid assignment to union data for directional parameter

[Sasha Levin]

From: Takashi Sakamoto <o-takashi@sakamocchi.jp>

[ Upstream commit 8304cf77c92038cd1c50c27b69d30be695cc8003 ]

Although the value of FDF is used just for outgoing stream, the assignment
to union member is done for both directions of stream. At present this
causes no issue because the value of same position is reassigned later for
opposite stream. However, it's better to add if statement.

Fixes: d3d10a4a1b19 ("ALSA: firewire-lib: use union for directional parameters")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/firewire/amdtp-am824.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 67d735e9a6a4..fea92e148790 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -82,7 +82,8 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (err < 0)
 		return err;
 
-	s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
+	if (s->direction == AMDTP_OUT_STREAM)
+		s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
 
 	p->pcm_channels = pcm_channels;
 	p->midi_ports = midi_ports;
-- 
2.25.1

[PATCH AUTOSEL 5.7 183/388] ASoC: SOF: core: fix error return code in sof_probe_continue()

[Sasha Levin]

From: Wei Yongjun <weiyongjun1@huawei.com>

[ Upstream commit 7d8785bc7adbb4dc5ba8ee06994107637848ded8 ]

Fix to return negative error code -ENOMEM from the IPC init error
handling case instead of 0, as done elsewhere in this function.

Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/core.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 91acfae7935c..74b438216216 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -176,6 +176,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev)
 	/* init the IPC */
 	sdev->ipc = snd_sof_ipc_init(sdev);
 	if (!sdev->ipc) {
+		ret = -ENOMEM;
 		dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret);
 		goto ipc_err;
 	}
-- 
2.25.1

[PATCH AUTOSEL 5.7 193/388] SoC: rsnd: add interrupt support for SSI BUSIF buffer

[Sasha Levin]

From: Yongbo Zhang <giraffesnn123@gmail.com>

[ Upstream commit 66c705d07d784fb6b4622c6e47b6acae357472db ]

SSI BUSIF buffer is possible to overflow or underflow, especially in a
hypervisor environment. If there is no interrupt support, it will eventually
lead to errors in pcm data.
This patch adds overflow and underflow interrupt support for SSI BUSIF buffer.

Reported-by: Chen Li <licheng0822@thundersoft.com>
Signed-off-by: Yongbo Zhang <giraffesnn123@gmail.com>
Tested-by: Chen Li <licheng0822@thundersoft.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sh/rcar/gen.c  |   8 +++
 sound/soc/sh/rcar/rsnd.h |   9 +++
 sound/soc/sh/rcar/ssi.c  | 145 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 162 insertions(+)

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index af19010b9d88..8bd49c8a9517 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
 		RSND_GEN_S_REG(SSI_SYS_STATUS5,	0x884),
 		RSND_GEN_S_REG(SSI_SYS_STATUS6,	0x888),
 		RSND_GEN_S_REG(SSI_SYS_STATUS7,	0x88c),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
+		RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c),
 		RSND_GEN_S_REG(HDMI0_SEL,	0x9e0),
 		RSND_GEN_S_REG(HDMI1_SEL,	0x9e4),
 
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index ea6cbaa9743e..d47608ff5fac 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -189,6 +189,14 @@ enum rsnd_reg {
 	SSI_SYS_STATUS5,
 	SSI_SYS_STATUS6,
 	SSI_SYS_STATUS7,
+	SSI_SYS_INT_ENABLE0,
+	SSI_SYS_INT_ENABLE1,
+	SSI_SYS_INT_ENABLE2,
+	SSI_SYS_INT_ENABLE3,
+	SSI_SYS_INT_ENABLE4,
+	SSI_SYS_INT_ENABLE5,
+	SSI_SYS_INT_ENABLE6,
+	SSI_SYS_INT_ENABLE7,
 	HDMI0_SEL,
 	HDMI1_SEL,
 	SSI9_BUSIF0_MODE,
@@ -237,6 +245,7 @@ enum rsnd_reg {
 #define SSI9_BUSIF_ADINR(i)	(SSI9_BUSIF0_ADINR + (i))
 #define SSI9_BUSIF_DALIGN(i)	(SSI9_BUSIF0_DALIGN + (i))
 #define SSI_SYS_STATUS(i)	(SSI_SYS_STATUS0 + (i))
+#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i))
 
 
 struct rsnd_priv;
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4a7d3413917f..47d5ddb526f2 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	u32 wsr		= ssi->wsr;
 	int width;
 	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+	int i;
+	u32 sys_int_enable = 0;
 
 	is_tdm		= rsnd_runtime_is_tdm(io);
 	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
@@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_mode = DIEN;		/* PIO : enable Data interrupt */
 	}
 
+	/* enable busif buffer over/under run interrupt. */
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE(i * 2));
+				sys_int_enable |= 0xf << (id * 4);
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE(i * 2),
+					       sys_int_enable);
+			}
+
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE((i * 2) + 1));
+				sys_int_enable |= 0xf << 4;
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE((i * 2) + 1),
+					       sys_int_enable);
+			}
+
+			break;
+		}
+	}
+
 init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
@@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
+	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+	int i;
+	u32 sys_int_enable = 0;
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
@@ -517,6 +559,38 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod,
 		ssi->wsr	= 0;
 	}
 
+	/* disable busif buffer over/under run interrupt. */
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+						SSI_SYS_INT_ENABLE(i * 2));
+				sys_int_enable &= ~(0xf << (id * 4));
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE(i * 2),
+					       sys_int_enable);
+			}
+
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				sys_int_enable = rsnd_mod_read(mod,
+					SSI_SYS_INT_ENABLE((i * 2) + 1));
+				sys_int_enable &= ~(0xf << 4);
+				rsnd_mod_write(mod,
+					       SSI_SYS_INT_ENABLE((i * 2) + 1),
+					       sys_int_enable);
+			}
+
+			break;
+		}
+	}
+
 	return 0;
 }
 
@@ -622,6 +696,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 			int enable)
 {
 	u32 val = 0;
+	int is_tdm, is_tdm_split;
+	int id = rsnd_mod_id(mod);
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	if (rsnd_is_gen1(priv))
 		return 0;
@@ -635,6 +714,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 	if (enable)
 		val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000;
 
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+		case 9:
+			val |= 0x0000ff00;
+			break;
+		}
+	}
+
 	rsnd_mod_write(mod, SSI_INT_ENABLE, val);
 
 	return 0;
@@ -651,6 +743,12 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod,
 	u32 status;
 	bool elapsed = false;
 	bool stop = false;
+	int id = rsnd_mod_id(mod);
+	int i;
+	int is_tdm, is_tdm_split;
+
+	is_tdm		= rsnd_runtime_is_tdm(io);
+	is_tdm_split	= rsnd_runtime_is_tdm_split(io);
 
 	spin_lock(&priv->lock);
 
@@ -672,6 +770,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod,
 		stop = true;
 	}
 
+	status = 0;
+
+	if (is_tdm || is_tdm_split) {
+		switch (id) {
+		case 0:
+		case 1:
+		case 2:
+		case 3:
+		case 4:
+			for (i = 0; i < 4; i++) {
+				status = rsnd_mod_read(mod,
+						       SSI_SYS_STATUS(i * 2));
+				status &= 0xf << (id * 4);
+
+				if (status) {
+					rsnd_dbg_irq_status(dev,
+						"%s err status : 0x%08x\n",
+						rsnd_mod_name(mod), status);
+					rsnd_mod_write(mod,
+						       SSI_SYS_STATUS(i * 2),
+						       0xf << (id * 4));
+					stop = true;
+					break;
+				}
+			}
+			break;
+		case 9:
+			for (i = 0; i < 4; i++) {
+				status = rsnd_mod_read(mod,
+						SSI_SYS_STATUS((i * 2) + 1));
+				status &= 0xf << 4;
+
+				if (status) {
+					rsnd_dbg_irq_status(dev,
+						"%s err status : 0x%08x\n",
+						rsnd_mod_name(mod), status);
+					rsnd_mod_write(mod,
+						SSI_SYS_STATUS((i * 2) + 1),
+						0xf << 4);
+					stop = true;
+					break;
+				}
+			}
+			break;
+		}
+	}
+
 	rsnd_ssi_status_clear(mod);
 rsnd_ssi_interrupt_out:
 	spin_unlock(&priv->lock);
-- 
2.25.1

[PATCH AUTOSEL 5.7 194/388] ASoC: ux500: mop500: Fix some refcounted resources issues

[Sasha Levin]

From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>

[ Upstream commit 4e8748fcaeec073e3ba794871ce86c545e4f961f ]

There are 2 issues here:
   - if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()'
     is a no-op because the 'mop500_dai_links' structure has not been
     initialized yet, so the referenced are not decremented
   - The reference stored in 'mop500_dai_links[i].codecs' is refcounted
     only once in the probe and must be decremented only once.

Fixes: 39013bd60e79 ("ASoC: Ux500: Dispose of device nodes correctly")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/ux500/mop500.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 2873e8e6f02b..cdae1190b930 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -63,10 +63,11 @@ static void mop500_of_node_put(void)
 {
 	int i;
 
-	for (i = 0; i < 2; i++) {
+	for (i = 0; i < 2; i++)
 		of_node_put(mop500_dai_links[i].cpus->of_node);
-		of_node_put(mop500_dai_links[i].codecs->of_node);
-	}
+
+	/* Both links use the same codec, which is refcounted only once */
+	of_node_put(mop500_dai_links[0].codecs->of_node);
 }
 
 static int mop500_of_probe(struct platform_device *pdev,
@@ -81,7 +82,9 @@ static int mop500_of_probe(struct platform_device *pdev,
 
 	if (!(msp_np[0] && msp_np[1] && codec_np)) {
 		dev_err(&pdev->dev, "Phandle missing or invalid\n");
-		mop500_of_node_put();
+		for (i = 0; i < 2; i++)
+			of_node_put(msp_np[i]);
+		of_node_put(codec_np);
 		return -EINVAL;
 	}
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 195/388] ASoC: ti: omap-mcbsp: Fix an error handling path in 'asoc_mcbsp_probe()'

[Sasha Levin]

From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>

[ Upstream commit 03990fd58d2b7c8f7d53e514ba9b8749fac260f9 ]

If an error occurs after the call to 'omap_mcbsp_init()', the reference to
'mcbsp->fclk' must be decremented, as already done in the remove function.

This can be achieved easily by using the devm_ variant of 'clk_get()'
when the reference is taken in 'omap_mcbsp_init()'

This fixes the leak in the probe and has the side effect to simplify both
the error handling path of 'omap_mcbsp_init()' and the remove function.

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Peter Ujfalusi <peter.ujflausi@ti.com>
Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/ti/omap-mcbsp.c | 8 ++------
 1 file changed, 2 insertions(+), 6 deletions(-)

diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 3d41ca2238d4..4f33ddb7b441 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -686,7 +686,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 	mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp,
 						SNDRV_PCM_STREAM_CAPTURE);
 
-	mcbsp->fclk = clk_get(&pdev->dev, "fck");
+	mcbsp->fclk = devm_clk_get(&pdev->dev, "fck");
 	if (IS_ERR(mcbsp->fclk)) {
 		ret = PTR_ERR(mcbsp->fclk);
 		dev_err(mcbsp->dev, "unable to get fck: %d\n", ret);
@@ -711,7 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		if (ret) {
 			dev_err(mcbsp->dev,
 				"Unable to create additional controls\n");
-			goto err_thres;
+			return ret;
 		}
 	}
 
@@ -724,8 +724,6 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 err_st:
 	if (mcbsp->pdata->buffer_size)
 		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-err_thres:
-	clk_put(mcbsp->fclk);
 	return ret;
 }
 
@@ -1442,8 +1440,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
 
 	omap_mcbsp_st_cleanup(pdev);
 
-	clk_put(mcbsp->fclk);
-
 	return 0;
 }
 
-- 
2.25.1

[PATCH AUTOSEL 5.7 209/388] ALSA: usb-audio: fixing upper volume limit for RME Babyface Pro routing crosspoints

[Sasha Levin]

From: Thomas Ebeling <penguins@bollie.de>

[ Upstream commit 47b4f5f5b65680fbef7a7a9a4796b35f38a6e43e ]

In my initial patch, these were set too low.

Fixes: 3e8f3bd04716 ("ALSA: usb-audio: RME Babyface Pro mixer patch")
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Link: https://lore.kernel.org/r/20200515114556.vtspnonzvp4xp44m@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/mixer_quirks.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index bdff8674942e..aad2683ff793 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2191,7 +2191,7 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer)
  * These devices exposes a couple of DSP functions via request to EP0.
  * Switches are available via control registers, while routing is controlled
  * by controlling the volume on each possible crossing point.
- * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with
+ * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 65536) with
  * 0dB being at dec. 32768.
  */
 enum {
@@ -2220,7 +2220,7 @@ enum {
 #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff
 #define SND_BBFPRO_MIXER_VAL_SHIFT 9
 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf
-#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB
+#define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB
 
 #define SND_BBFPRO_USBREQ_CTL_REG1 0x10
 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17
-- 
2.25.1

[PATCH AUTOSEL 5.7 210/388] ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback

[Sasha Levin]

From: Erwin Burema <e.burema@gmail.com>

[ Upstream commit 10ce77e4817fef99e1166be7e6685a80c63bf77f ]

For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.

This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")

Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series")
Signed-off-by: Erwin Burema <e.burema@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751
Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/usb/card.h     |   1 +
 sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++-
 sound/usb/pcm.c      |   5 ++
 3 files changed, 197 insertions(+), 4 deletions(-)

diff --git a/sound/usb/card.h b/sound/usb/card.h
index 820e564656ed..d6219fba9699 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -108,6 +108,7 @@ struct snd_usb_endpoint {
 	int iface, altsetting;
 	int skip_packets;		/* quirks for devices to ignore the first n packets
 					   in a stream */
+	bool is_implicit_feedback;      /* This endpoint is used as implicit feedback */
 
 	spinlock_t lock;
 	struct list_head list;
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 50104f658ed4..9bea7d3f99f8 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 
 	list_add_tail(&ep->list, &chip->ep_list);
 
+	ep->is_implicit_feedback = 0;
+
 __exit_unlock:
 	mutex_unlock(&chip->mutex);
 
@@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force)
 	ep->nurbs = 0;
 }
 
+/*
+ * Check data endpoint for format differences
+ */
+static bool check_ep_params(struct snd_usb_endpoint *ep,
+			      snd_pcm_format_t pcm_format,
+			      unsigned int channels,
+			      unsigned int period_bytes,
+			      unsigned int frames_per_period,
+			      unsigned int periods_per_buffer,
+			      struct audioformat *fmt,
+			      struct snd_usb_endpoint *sync_ep)
+{
+	unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb;
+	unsigned int max_packs_per_period, urbs_per_period, urb_packs;
+	unsigned int max_urbs;
+	int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
+	int tx_length_quirk = (ep->chip->tx_length_quirk &&
+			       usb_pipeout(ep->pipe));
+	bool ret = 1;
+
+	if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
+		/*
+		 * When operating in DSD DOP mode, the size of a sample frame
+		 * in hardware differs from the actual physical format width
+		 * because we need to make room for the DOP markers.
+		 */
+		frame_bits += channels << 3;
+	}
+
+	ret = ret && (ep->datainterval == fmt->datainterval);
+	ret = ret && (ep->stride == frame_bits >> 3);
+
+	switch (pcm_format) {
+	case SNDRV_PCM_FORMAT_U8:
+		ret = ret && (ep->silence_value == 0x80);
+		break;
+	case SNDRV_PCM_FORMAT_DSD_U8:
+	case SNDRV_PCM_FORMAT_DSD_U16_LE:
+	case SNDRV_PCM_FORMAT_DSD_U32_LE:
+	case SNDRV_PCM_FORMAT_DSD_U16_BE:
+	case SNDRV_PCM_FORMAT_DSD_U32_BE:
+		ret = ret && (ep->silence_value == 0x69);
+		break;
+	default:
+		ret = ret && (ep->silence_value == 0);
+	}
+
+	/* assume max. frequency is 50% higher than nominal */
+	ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1));
+	/* Round up freqmax to nearest integer in order to calculate maximum
+	 * packet size, which must represent a whole number of frames.
+	 * This is accomplished by adding 0x0.ffff before converting the
+	 * Q16.16 format into integer.
+	 * In order to accurately calculate the maximum packet size when
+	 * the data interval is more than 1 (i.e. ep->datainterval > 0),
+	 * multiply by the data interval prior to rounding. For instance,
+	 * a freqmax of 41 kHz will result in a max packet size of 6 (5.125)
+	 * frames with a data interval of 1, but 11 (10.25) frames with a
+	 * data interval of 2.
+	 * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the
+	 * maximum datainterval value of 3, at USB full speed, higher for
+	 * USB high speed, noting that ep->freqmax is in units of
+	 * frames per packet in Q16.16 format.)
+	 */
+	maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
+			 (frame_bits >> 3);
+	if (tx_length_quirk)
+		maxsize += sizeof(__le32); /* Space for length descriptor */
+	/* but wMaxPacketSize might reduce this */
+	if (ep->maxpacksize && ep->maxpacksize < maxsize) {
+		/* whatever fits into a max. size packet */
+		unsigned int data_maxsize = maxsize = ep->maxpacksize;
+
+		if (tx_length_quirk)
+			/* Need to remove the length descriptor to calc freq */
+			data_maxsize -= sizeof(__le32);
+		ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3))
+				<< (16 - ep->datainterval));
+	}
+
+	if (ep->fill_max)
+		ret = ret && (ep->curpacksize == ep->maxpacksize);
+	else
+		ret = ret && (ep->curpacksize == maxsize);
+
+	if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) {
+		packs_per_ms = 8 >> ep->datainterval;
+		max_packs_per_urb = MAX_PACKS_HS;
+	} else {
+		packs_per_ms = 1;
+		max_packs_per_urb = MAX_PACKS;
+	}
+	if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep))
+		max_packs_per_urb = min(max_packs_per_urb,
+					1U << sync_ep->syncinterval);
+	max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval);
+
+	/*
+	 * Capture endpoints need to use small URBs because there's no way
+	 * to tell in advance where the next period will end, and we don't
+	 * want the next URB to complete much after the period ends.
+	 *
+	 * Playback endpoints with implicit sync much use the same parameters
+	 * as their corresponding capture endpoint.
+	 */
+	if (usb_pipein(ep->pipe) ||
+			snd_usb_endpoint_implicit_feedback_sink(ep)) {
+
+		urb_packs = packs_per_ms;
+		/*
+		 * Wireless devices can poll at a max rate of once per 4ms.
+		 * For dataintervals less than 5, increase the packet count to
+		 * allow the host controller to use bursting to fill in the
+		 * gaps.
+		 */
+		if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+			int interval = ep->datainterval;
+
+			while (interval < 5) {
+				urb_packs <<= 1;
+				++interval;
+			}
+		}
+		/* make capture URBs <= 1 ms and smaller than a period */
+		urb_packs = min(max_packs_per_urb, urb_packs);
+		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+			urb_packs >>= 1;
+		ret = ret && (ep->nurbs == MAX_URBS);
+
+	/*
+	 * Playback endpoints without implicit sync are adjusted so that
+	 * a period fits as evenly as possible in the smallest number of
+	 * URBs.  The total number of URBs is adjusted to the size of the
+	 * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits.
+	 */
+	} else {
+		/* determine how small a packet can be */
+		minsize = (ep->freqn >> (16 - ep->datainterval)) *
+				(frame_bits >> 3);
+		/* with sync from device, assume it can be 12% lower */
+		if (sync_ep)
+			minsize -= minsize >> 3;
+		minsize = max(minsize, 1u);
+
+		/* how many packets will contain an entire ALSA period? */
+		max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize);
+
+		/* how many URBs will contain a period? */
+		urbs_per_period = DIV_ROUND_UP(max_packs_per_period,
+				max_packs_per_urb);
+		/* how many packets are needed in each URB? */
+		urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period);
+
+		/* limit the number of frames in a single URB */
+		ret = ret && (ep->max_urb_frames ==
+			DIV_ROUND_UP(frames_per_period, urbs_per_period));
+
+		/* try to use enough URBs to contain an entire ALSA buffer */
+		max_urbs = min((unsigned) MAX_URBS,
+				MAX_QUEUE * packs_per_ms / urb_packs);
+		ret = ret && (ep->nurbs == min(max_urbs,
+				urbs_per_period * periods_per_buffer));
+	}
+
+	ret = ret && (ep->datainterval == fmt->datainterval);
+	ret = ret && (ep->maxpacksize == fmt->maxpacksize);
+	ret = ret &&
+		(ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX));
+
+	return ret;
+}
+
 /*
  * configure a data endpoint
  */
@@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 	int err;
 
 	if (ep->use_count != 0) {
-		usb_audio_warn(ep->chip,
-			 "Unable to change format on ep #%x: already in use\n",
-			 ep->ep_num);
-		return -EBUSY;
+		bool check = ep->is_implicit_feedback &&
+			check_ep_params(ep, pcm_format,
+					     channels, period_bytes,
+					     period_frames, buffer_periods,
+					     fmt, sync_ep);
+
+		if (!check) {
+			usb_audio_warn(ep->chip,
+				"Unable to change format on ep #%x: already in use\n",
+				ep->ep_num);
+			return -EBUSY;
+		}
+
+		usb_audio_dbg(ep->chip,
+			      "Ep #%x already in use as implicit feedback but format not changed\n",
+			      ep->ep_num);
+		return 0;
 	}
 
 	/* release old buffers, if any */
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index b50965ab3b3a..d61c2f1095b5 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -404,6 +404,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	if (!subs->sync_endpoint)
 		return -EINVAL;
 
+	subs->sync_endpoint->is_implicit_feedback = 1;
+
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
 	return 1;
@@ -502,12 +504,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 						   implicit_fb ?
 							SND_USB_ENDPOINT_TYPE_DATA :
 							SND_USB_ENDPOINT_TYPE_SYNC);
+
 	if (!subs->sync_endpoint) {
 		if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
 			return 0;
 		return -EINVAL;
 	}
 
+	subs->sync_endpoint->is_implicit_feedback = implicit_fb;
+
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
 	return 0;
-- 
2.25.1

[PATCH AUTOSEL 5.7 217/388] ASoC: codecs: rt*-sdw: fix memory leak in set_sdw_stream()

[Sasha Levin]

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

[ Upstream commit 07b542fe831cbefce163ad1b3aa7292c8a6332b8 ]

Now that the sdw_stream is allocated in machine driver,
set_sdw_stream() is also called with a NULL argument during the
dailink shutdown.

In this case, the drivers should not allocate any memory, and just
return.

Detected with KASAN/kmemleak.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Cc: Oder Chiou <oder_chiou@realtek.com>
Cc: Shuming Fan <shumingf@realtek.com>
Cc: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20200515211531.11416-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/rt1308-sdw.c | 3 +++
 sound/soc/codecs/rt5682.c     | 3 +++
 sound/soc/codecs/rt700.c      | 3 +++
 sound/soc/codecs/rt711.c      | 3 +++
 sound/soc/codecs/rt715.c      | 3 +++
 5 files changed, 15 insertions(+)

diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index a5a7e46de246..a7f45191364d 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -482,6 +482,9 @@ static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
 {
 	struct sdw_stream_data *stream;
 
+	if (!sdw_stream)
+		return 0;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index d36f560ad7a8..c4892af14850 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -2958,6 +2958,9 @@ static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
 {
 	struct sdw_stream_data *stream;
 
+	if (!sdw_stream)
+		return 0;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c
index ff68f0e4f629..687ac2153666 100644
--- a/sound/soc/codecs/rt700.c
+++ b/sound/soc/codecs/rt700.c
@@ -860,6 +860,9 @@ static int rt700_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
 {
 	struct sdw_stream_data *stream;
 
+	if (!sdw_stream)
+		return 0;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c
index 2daed7692a3b..65b59dbfb43c 100644
--- a/sound/soc/codecs/rt711.c
+++ b/sound/soc/codecs/rt711.c
@@ -906,6 +906,9 @@ static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
 {
 	struct sdw_stream_data *stream;
 
+	if (!sdw_stream)
+		return 0;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c
index 2cbc57b16b13..099c8bd20006 100644
--- a/sound/soc/codecs/rt715.c
+++ b/sound/soc/codecs/rt715.c
@@ -530,6 +530,9 @@ static int rt715_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
 
 	struct sdw_stream_data *stream;
 
+	if (!sdw_stream)
+		return 0;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
-- 
2.25.1

[PATCH AUTOSEL 5.7 232/388] 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 08f4ae964b02..fbfd53874b47 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -742,6 +742,18 @@ 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 5.7 279/388] 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 a495d1050d49..e30b66b94bf6 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -482,6 +482,7 @@ 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 5.7 294/388] ASoC: dapm: Move dai_link widgets to runtime to fix use after free

[Sasha Levin]

From: Charles Keepax <ckeepax@opensource.cirrus.com>

[ Upstream commit f4aa5e214eeaf7f1c7f157526a5aa29784cb6a1f ]

The newly added CODEC to CODEC DAI link widget pointers in
snd_soc_dai_link are better placed in snd_soc_pcm_runtime.
snd_soc_dai_link is really intended for static configuration of
the DAI, and the runtime for dynamic data.  The snd_soc_dai_link
structures are not destroyed if the card is unbound. The widgets
are cleared up on unbind, however if the card is rebound as the
snd_soc_dai_link structures are reused these pointers will be left at
their old values, causing access to freed memory.

Fixes: 595571cca4de ("ASoC: dapm: Fix regression introducing multiple copies of DAI widgets")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200526161930.30759-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 include/sound/soc.h  |  6 +++---
 sound/soc/soc-dapm.c | 12 ++++++------
 2 files changed, 9 insertions(+), 9 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index e0371e70242d..8e480efeda2a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -790,9 +790,6 @@ struct snd_soc_dai_link {
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
 
-	struct snd_soc_dapm_widget *playback_widget;
-	struct snd_soc_dapm_widget *capture_widget;
-
 	unsigned int dai_fmt;           /* format to set on init */
 
 	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
@@ -1156,6 +1153,9 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_dai **cpu_dais;
 	unsigned int num_cpus;
 
+	struct snd_soc_dapm_widget *playback_widget;
+	struct snd_soc_dapm_widget *capture_widget;
+
 	struct delayed_work delayed_work;
 	void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
 #ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index e2632841b321..c0aa64ff8e32 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4340,16 +4340,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card,
 	codec = codec_dai->playback_widget;
 
 	if (playback_cpu && codec) {
-		if (dai_link->params && !dai_link->playback_widget) {
+		if (dai_link->params && !rtd->playback_widget) {
 			substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
 			dai = snd_soc_dapm_new_dai(card, substream, "playback");
 			if (IS_ERR(dai))
 				goto capture;
-			dai_link->playback_widget = dai;
+			rtd->playback_widget = dai;
 		}
 
 		dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu,
-					dai_link->playback_widget,
+					rtd->playback_widget,
 					codec_dai, codec);
 	}
 
@@ -4358,16 +4358,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card,
 	codec = codec_dai->capture_widget;
 
 	if (codec && capture_cpu) {
-		if (dai_link->params && !dai_link->capture_widget) {
+		if (dai_link->params && !rtd->capture_widget) {
 			substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
 			dai = snd_soc_dapm_new_dai(card, substream, "capture");
 			if (IS_ERR(dai))
 				return;
-			dai_link->capture_widget = dai;
+			rtd->capture_widget = dai;
 		}
 
 		dapm_connect_dai_routes(&card->dapm, codec_dai, codec,
-					dai_link->capture_widget,
+					rtd->capture_widget,
 					cpu_dai, capture_cpu);
 	}
 }
-- 
2.25.1

[PATCH AUTOSEL 5.7 306/388] 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 e7178817d7a7..1ee10eafe3e6 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -252,6 +252,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	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 5.7 307/388] ASoC: SOF: Intel: hda: fix generic hda codec support

[Sasha Levin]

From: Kai Vehmanen <kai.vehmanen@linux.intel.com>

[ Upstream commit 89d73ccab20a684d8446cea4d8ac6a2608c8d390 ]

Add support for using generic codec driver with SOF. Generic driver
is used if:
 - snd_sof_intel_hda_common.hda_model="generic" is set, or
 - fallback if no other codec driver is found

The implementation is aligned with snd-hda-intel driver, and fixes audio
support for systems like Acer Swift 3 SF314-57G, on which this issue was
originally reported.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Hui Wang <hui.wang@canonical.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/1807
BugLink: https://bugs.launchpad.net/bugs/1877757
Link: https://lore.kernel.org/r/20200529160358.12134-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/hda-codec.c | 51 +++++++++++++++++++++++++++++----
 1 file changed, 45 insertions(+), 6 deletions(-)

diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 3041fbbb010a..ea021db697b8 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -24,19 +24,44 @@
 #define IDISP_VID_INTEL	0x80860000
 
 /* load the legacy HDA codec driver */
-static int hda_codec_load_module(struct hda_codec *codec)
+static int request_codec_module(struct hda_codec *codec)
 {
 #ifdef MODULE
 	char alias[MODULE_NAME_LEN];
-	const char *module = alias;
+	const char *mod = NULL;
 
-	snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias));
-	dev_dbg(&codec->core.dev, "loading codec module: %s\n", module);
-	request_module(module);
+	switch (codec->probe_id) {
+	case HDA_CODEC_ID_GENERIC:
+#if IS_MODULE(CONFIG_SND_HDA_GENERIC)
+		mod = "snd-hda-codec-generic";
 #endif
+		break;
+	default:
+		snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias));
+		mod = alias;
+		break;
+	}
+
+	if (mod) {
+		dev_dbg(&codec->core.dev, "loading codec module: %s\n", mod);
+		request_module(mod);
+	}
+#endif /* MODULE */
 	return device_attach(hda_codec_dev(codec));
 }
 
+static int hda_codec_load_module(struct hda_codec *codec)
+{
+	int ret = request_codec_module(codec);
+
+	if (ret <= 0) {
+		codec->probe_id = HDA_CODEC_ID_GENERIC;
+		ret = request_codec_module(codec);
+	}
+
+	return ret;
+}
+
 /* enable controller wake up event for all codecs with jack connectors */
 void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
 {
@@ -78,6 +103,13 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) {}
 EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC);
 EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC);
 
+#if IS_ENABLED(CONFIG_SND_HDA_GENERIC)
+#define is_generic_config(bus) \
+	((bus)->modelname && !strcmp((bus)->modelname, "generic"))
+#else
+#define is_generic_config(x)	0
+#endif
+
 /* probe individual codec */
 static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 			   bool hda_codec_use_common_hdmi)
@@ -87,6 +119,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 #endif
 	struct hda_bus *hbus = sof_to_hbus(sdev);
 	struct hdac_device *hdev;
+	struct hda_codec *codec;
 	u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) |
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	u32 resp = -1;
@@ -108,6 +141,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 
 	hda_priv->codec.bus = hbus;
 	hdev = &hda_priv->codec.core;
+	codec = &hda_priv->codec;
 
 	ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev);
 	if (ret < 0)
@@ -122,6 +156,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 		hda_priv->need_display_power = true;
 	}
 
+	if (is_generic_config(hbus))
+		codec->probe_id = HDA_CODEC_ID_GENERIC;
+	else
+		codec->probe_id = 0;
+
 	/*
 	 * if common HDMI codec driver is not used, codec load
 	 * is skipped here and hdac_hdmi is used instead
@@ -129,7 +168,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 	if (hda_codec_use_common_hdmi ||
 	    (resp & 0xFFFF0000) != IDISP_VID_INTEL) {
 		hdev->type = HDA_DEV_LEGACY;
-		ret = hda_codec_load_module(&hda_priv->codec);
+		ret = hda_codec_load_module(codec);
 		/*
 		 * handle ret==0 (no driver bound) as an error, but pass
 		 * other return codes without modification
-- 
2.25.1

[PATCH AUTOSEL 5.7 362/388] ASoC: soc-pcm: dpcm: fix playback/capture checks

[Sasha Levin]

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

[ Upstream commit b73287f0b0745961b14e5ebcce92cc8ed24d4d52 ]

Recent changes in the ASoC core prevent multi-cpu BE dailinks from
being used. DPCM does support multi-cpu DAIs for BE Dailinks, but not
for FE.

Handle the FE checks first, and make sure all DAIs support the same
capabilities within the same dailink.

Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
BugLink: https://github.com/thesofproject/linux/issues/2031
Link: https://lore.kernel.org/r/20200608194415.4663-2-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-pcm.c | 44 ++++++++++++++++++++++++++++++++++----------
 1 file changed, 34 insertions(+), 10 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 1f302de44052..39ce61c5b874 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2908,20 +2908,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	struct snd_pcm *pcm;
 	char new_name[64];
 	int ret = 0, playback = 0, capture = 0;
+	int stream;
 	int i;
 
+	if (rtd->dai_link->dynamic && rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"DPCM doesn't support Multi CPU for Front-Ends yet\n");
+		return -EINVAL;
+	}
+
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-		if (rtd->num_cpus > 1) {
-			dev_err(rtd->dev,
-				"DPCM doesn't support Multi CPU yet\n");
-			return -EINVAL;
+		if (rtd->dai_link->dpcm_playback) {
+			stream = SNDRV_PCM_STREAM_PLAYBACK;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support playback\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			playback = 1;
+		}
+		if (rtd->dai_link->dpcm_capture) {
+			stream = SNDRV_PCM_STREAM_CAPTURE;
+
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				if (!snd_soc_dai_stream_valid(cpu_dai,
+							      stream)) {
+					dev_err(rtd->card->dev,
+						"CPU DAI %s for rtd %s does not support capture\n",
+						cpu_dai->name,
+						rtd->dai_link->stream_name);
+					return -EINVAL;
+				}
+			capture = 1;
 		}
-
-		playback = rtd->dai_link->dpcm_playback &&
-			   snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK);
-		capture = rtd->dai_link->dpcm_capture &&
-			  snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE);
 	} else {
 		/* Adapt stream for codec2codec links */
 		int cpu_capture = rtd->dai_link->params ?
-- 
2.25.1

[PATCH AUTOSEL 5.7 363/388] 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 843b8b1c89d4..e5433e8fcf19 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1720,9 +1720,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			dai_link->platforms->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 5.7 364/388] ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags

[Sasha Levin]

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

[ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ]

With additional checks on dailinks, we see errors such as

[ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd
NoCodec-6 does not support playback

It's not clear why we set the dpcm_playback and dpcm_capture flags
unconditionally, add a check on number of channels for each direction
to avoid invalid configurations.

Fixes: 8017b8fd37bf5e ('ASoC: SOF: Add Nocodec machine driver support')
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/20200608194415.4663-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/nocodec.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index 2233146386cc..71cf5f9db79d 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev,
 		links[i].platforms->name = dev_name(dev);
 		links[i].codecs->dai_name = "snd-soc-dummy-dai";
 		links[i].codecs->name = "snd-soc-dummy";
-		links[i].dpcm_playback = 1;
-		links[i].dpcm_capture = 1;
+		if (ops->drv[i].playback.channels_min)
+			links[i].dpcm_playback = 1;
+		if (ops->drv[i].capture.channels_min)
+			links[i].dpcm_capture = 1;
 	}
 
 	card->dai_link = links;
-- 
2.25.1

[PATCH AUTOSEL 5.7 365/388] 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 fbfd53874b47..5c1a5e2aff6f 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -754,6 +754,18 @@ 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 5.7 366/388] 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 6ba1849a77b0..e2e1d5b03b38 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3625,6 +3625,12 @@ 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,
@@ -3708,6 +3714,14 @@ 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

Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 494 bytes --]

On Wed, Jun 17, 2020 at 09:01:41PM -0400, Sasha Levin wrote:
> 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.

This is a new feature not a bugfix.

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Re: [PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set

[Mark Brown]

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On Wed, Jun 17, 2020 at 09:02:32PM -0400, Sasha Levin wrote:
> From: Daniel Baluta <daniel.baluta@nxp.com>
> 
> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ]
> 
> This provides a better separation between runtime and PM sleep
> callbacks.
> 
> Only do nothing if given runtime flag is set and calback is not set.
> 
> With the current implementation, if PM sleep callback is set but runtime
> callback is not set then at runtime resume we reload the firmware even
> if we do not support runtime resume callback.

This doesn't look like a bugfix, just an optimization?

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Re: [PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983

[Mark Brown]

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On Wed, Jun 17, 2020 at 09:03:47PM -0400, Sasha Levin wrote:
> From: Wei Li <liwei391@huawei.com>
> 
> [ Upstream commit c1c050ee74d67aeb879fd38e3a07139d7fdb79f4 ]
> 
> As these two drivers support I2C and SPI, we should add the SND_SOC_I2C_AND_SPI
> dependency instead.

This is purely about build testing, are you sure this is stable
material?

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Re: [PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set

[Daniel Baluta]

On 6/18/20 2:01 PM, Mark Brown wrote:
> On Wed, Jun 17, 2020 at 09:02:32PM -0400, Sasha Levin wrote:
>> From: Daniel Baluta <daniel.baluta@nxp.com>
>>
>> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ]
>>
>> This provides a better separation between runtime and PM sleep
>> callbacks.
>>
>> Only do nothing if given runtime flag is set and calback is not set.
>>
>> With the current implementation, if PM sleep callback is set but runtime
>> callback is not set then at runtime resume we reload the firmware even
>> if we do not support runtime resume callback.
> This doesn't look like a bugfix, just an optimization?

Indeed can be seen as an optimization, but it does unexpected things 
which can cause trouble

and weird behavior for people not familiar with the matter.

For example, as explained in the commit message if you only provide

System PM handler but not runtime PM handler, then the DSP will be resetted

even if this is not the intention.

Re: [PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set

[Mark Brown]

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On Thu, Jun 18, 2020 at 02:44:18PM +0300, Daniel Baluta wrote:

> Indeed can be seen as an optimization, but it does unexpected things which
> can cause trouble

> and weird behavior for people not familiar with the matter.

> For example, as explained in the commit message if you only provide

> System PM handler but not runtime PM handler, then the DSP will be resetted

> even if this is not the intention.

The user shouldn't be thinking about if the DSP is reset during runtime
PM, or if it's being reset...

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Re: [PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set

[Pierre-Louis Bossart]

On 6/18/20 6:44 AM, Daniel Baluta wrote:
> On 6/18/20 2:01 PM, Mark Brown wrote:
>> On Wed, Jun 17, 2020 at 09:02:32PM -0400, Sasha Levin wrote:
>>> From: Daniel Baluta <daniel.baluta@nxp.com>
>>>
>>> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ]
>>>
>>> This provides a better separation between runtime and PM sleep
>>> callbacks.
>>>
>>> Only do nothing if given runtime flag is set and calback is not set.
>>>
>>> With the current implementation, if PM sleep callback is set but runtime
>>> callback is not set then at runtime resume we reload the firmware even
>>> if we do not support runtime resume callback.
>> This doesn't look like a bugfix, just an optimization?
> 
> Indeed can be seen as an optimization, but it does unexpected things 
> which can cause trouble
> 
> and weird behavior for people not familiar with the matter.
> 
> For example, as explained in the commit message if you only provide
> 
> System PM handler but not runtime PM handler, then the DSP will be resetted
> 
> even if this is not the intention.

I think it's a bug fix for Intel legacy platforms (Baytrail, Broadwell) 
where runtime_pm isn't supported. However the additional fixes for 
system suspend/resume were only provided for 5.8, so this patch in 
isolation will not do much for those platforms. Put differently, even if 
this patch is applied to 5.7 suspend/resume would still not work for 
Baytrail/Broadwell.
Daniel, your call if you need this for i.MX?

Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Sasha Levin]

On Thu, Jun 18, 2020 at 12:00:23PM +0100, Mark Brown wrote:
>On Wed, Jun 17, 2020 at 09:01:41PM -0400, Sasha Levin wrote:
>> 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.
>
>This is a new feature not a bugfix.

I saw this patch more as a hardware quirk.

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Mark Brown]

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On Thu, Jun 18, 2020 at 10:30:46AM -0400, Sasha Levin wrote:
> On Thu, Jun 18, 2020 at 12:00:23PM +0100, Mark Brown wrote:
> > On Wed, Jun 17, 2020 at 09:01:41PM -0400, Sasha Levin wrote:
> > > 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.

> > This is a new feature not a bugfix.

> I saw this patch more as a hardware quirk.

Pretty much any DT property is a hardware quirk :(

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Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Sasha Levin]

On Thu, Jun 18, 2020 at 03:39:30PM +0100, Mark Brown wrote:
>On Thu, Jun 18, 2020 at 10:30:46AM -0400, Sasha Levin wrote:
>> On Thu, Jun 18, 2020 at 12:00:23PM +0100, Mark Brown wrote:
>> > On Wed, Jun 17, 2020 at 09:01:41PM -0400, Sasha Levin wrote:
>> > > 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.
>
>> > This is a new feature not a bugfix.
>
>> I saw this patch more as a hardware quirk.
>
>Pretty much any DT property is a hardware quirk :(

Which is why we're taking most of them :)

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983

[Sasha Levin]

On Thu, Jun 18, 2020 at 12:02:58PM +0100, Mark Brown wrote:
>On Wed, Jun 17, 2020 at 09:03:47PM -0400, Sasha Levin wrote:
>> From: Wei Li <liwei391@huawei.com>
>>
>> [ Upstream commit c1c050ee74d67aeb879fd38e3a07139d7fdb79f4 ]
>>
>> As these two drivers support I2C and SPI, we should add the SND_SOC_I2C_AND_SPI
>> dependency instead.
>
>This is purely about build testing, are you sure this is stable
>material?

Is this not something that can happen in practice?

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983

[Mark Brown]

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On Sun, Jun 21, 2020 at 07:34:53PM -0400, Sasha Levin wrote:
> On Thu, Jun 18, 2020 at 12:02:58PM +0100, Mark Brown wrote:

> > This is purely about build testing, are you sure this is stable
> > material?

> Is this not something that can happen in practice?

Not outside of build testing.

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Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Mark Brown]

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On Sun, Jun 21, 2020 at 07:33:52PM -0400, Sasha Levin wrote:
> On Thu, Jun 18, 2020 at 03:39:30PM +0100, Mark Brown wrote:
> > On Thu, Jun 18, 2020 at 10:30:46AM -0400, Sasha Levin wrote:
> > > On Thu, Jun 18, 2020 at 12:00:23PM +0100, Mark Brown wrote:

> > > > This is a new feature not a bugfix.

> > > I saw this patch more as a hardware quirk.

> > Pretty much any DT property is a hardware quirk :(

> Which is why we're taking most of them :)

That's concerning - please don't do this.  It's not what stable is
expected to be and there's no guarantee that you're getting all the
changes required to actually make things work.

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Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Sasha Levin]

On Mon, Jun 22, 2020 at 12:23:21PM +0100, Mark Brown wrote:
>On Sun, Jun 21, 2020 at 07:33:52PM -0400, Sasha Levin wrote:
>> On Thu, Jun 18, 2020 at 03:39:30PM +0100, Mark Brown wrote:
>> > On Thu, Jun 18, 2020 at 10:30:46AM -0400, Sasha Levin wrote:
>> > > On Thu, Jun 18, 2020 at 12:00:23PM +0100, Mark Brown wrote:
>
>> > > > This is a new feature not a bugfix.
>
>> > > I saw this patch more as a hardware quirk.
>
>> > Pretty much any DT property is a hardware quirk :(
>
>> Which is why we're taking most of them :)
>
>That's concerning - please don't do this.  It's not what stable is
>expected to be and there's no guarantee that you're getting all the
>changes required to actually make things work.

How come? This is one of the things stable rules explicitly call for:
"New device IDs and quirks are also accepted".

If we're missing anything, the solution is to make sure we stop missing
it rather than not take anything to begin with :)

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983

[Sasha Levin]

On Mon, Jun 22, 2020 at 11:18:30AM +0100, Mark Brown wrote:
>On Sun, Jun 21, 2020 at 07:34:53PM -0400, Sasha Levin wrote:
>> On Thu, Jun 18, 2020 at 12:02:58PM +0100, Mark Brown wrote:
>
>> > This is purely about build testing, are you sure this is stable
>> > material?
>
>> Is this not something that can happen in practice?
>
>Not outside of build testing.

Okay, I'll drop it.

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Mark Brown]

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On Mon, Jun 22, 2020 at 08:31:18AM -0400, Sasha Levin wrote:
> On Mon, Jun 22, 2020 at 12:23:21PM +0100, Mark Brown wrote:

> > That's concerning - please don't do this.  It's not what stable is
> > expected to be and there's no guarantee that you're getting all the
> > changes required to actually make things work.

> How come? This is one of the things stable rules explicitly call for:
> "New device IDs and quirks are also accepted".

I would expect that to be data only additions, I would not expect that
to be adding new code.

> If we're missing anything, the solution is to make sure we stop missing
> it rather than not take anything to begin with :)

It would be much better to not have to watch stable constantly like we
currently do - we're seeing people report breakage often enough to be a
concern as things are, we don't need to be trying to pile extra stuff in
there because there's some keywords in a changelog or whatever.  The
testing coverage for drivers is weak, increasing the change rate puts
more stress on that.

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Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Sasha Levin]

On Mon, Jun 22, 2020 at 02:27:57PM +0100, Mark Brown wrote:
>On Mon, Jun 22, 2020 at 08:31:18AM -0400, Sasha Levin wrote:
>> On Mon, Jun 22, 2020 at 12:23:21PM +0100, Mark Brown wrote:
>
>> > That's concerning - please don't do this.  It's not what stable is
>> > expected to be and there's no guarantee that you're getting all the
>> > changes required to actually make things work.
>
>> How come? This is one of the things stable rules explicitly call for:
>> "New device IDs and quirks are also accepted".
>
>I would expect that to be data only additions, I would not expect that
>to be adding new code.

These come hand in hand. Take a look at the more complex cases such as
sound/pci/hda/patch_*

>> If we're missing anything, the solution is to make sure we stop missing
>> it rather than not take anything to begin with :)
>
>It would be much better to not have to watch stable constantly like we
>currently do - we're seeing people report breakage often enough to be a
>concern as things are, we don't need to be trying to pile extra stuff in
>there because there's some keywords in a changelog or whatever.  The
>testing coverage for drivers is weak, increasing the change rate puts
>more stress on that.

Shouldn't we instead improve testing here? nvidia for example already
provides Tegra testing for stable releases, if the coverage isn't
sufficient then let's work on making it better.

-- 
Thanks,
Sasha

Re: [PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property

[Mark Brown]

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On Mon, Jun 22, 2020 at 10:44:02AM -0400, Sasha Levin wrote:
> On Mon, Jun 22, 2020 at 02:27:57PM +0100, Mark Brown wrote:
> > On Mon, Jun 22, 2020 at 08:31:18AM -0400, Sasha Levin wrote:

> > > How come? This is one of the things stable rules explicitly call for:
> > > "New device IDs and quirks are also accepted".

> > I would expect that to be data only additions, I would not expect that
> > to be adding new code.

> These come hand in hand. Take a look at the more complex cases such as
> sound/pci/hda/patch_*

There are more complex cases, I'm just not sure how good an idea
backporting them.

> > It would be much better to not have to watch stable constantly like we
> > currently do - we're seeing people report breakage often enough to be a
> > concern as things are, we don't need to be trying to pile extra stuff in
> > there because there's some keywords in a changelog or whatever.  The
> > testing coverage for drivers is weak, increasing the change rate puts
> > more stress on that.

> Shouldn't we instead improve testing here? nvidia for example already
> provides Tegra testing for stable releases, if the coverage isn't
> sufficient then let's work on making it better.

Obviously it'd be good to improve the test coverage, but I think that's
something that needs doing before backporting loads of stuff rather than
after.  For this automated stuff I'd much rather see positive
confirmation that the change had been tested on relevant systems (not
just something with a similar SoC), especially on the edges where we're
getting to board specific things.

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