[GIT PULL] sound fixes for 4.4-rc1

From: Takashi Iwai
Date: Sat Nov 14 2015 - 05:35:34 EST


Linus,

please pull sound fixes for v4.4-rc1 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-fix-4.4-rc1

The topmost commit is 2db1a57986d37653583e67ccbf13082aadc8f25d

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

sound fixes for 4.4-rc1

Here are a collection of small fixes tha have been gathered for
4.4-rc1. The only significant changes are those in PCI drivers
Kconfig, to use "depends on" instead of "select" for CONFIG_ZONE_DMA.
A reverse select is often more user-friendly, but in this case, it
makes hard to manage with the conflict with ZONE_DEVICE, so changed in
such a way for now.

Others are all small fixes and quirks: an error check in soundcore
reigster_chrdev(), HD-audio HDMI/DP phantom jack fix, Intel Broxton DP
quirk, USB-audio DSD device quirk, some constifications, etc.

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

Alexey Khoroshilov (1):
sound: fix check for error condition of register_chrdev()

Dan Williams (1):
ALSA: pci: depend on ZONE_DMA

Julia Lawall (1):
ALSA: ctxfi: constify rsc ops structures

Jurgen Kramer (1):
ALSA: usb: Add native DSD support for Aune X1S

Lu, Han (1):
ALSA: hda/hdmi - apply Skylake fix-ups to Broxton display codec

Takashi Iwai (1):
ALSA: hda - Simplify phantom jack handling for HDMI/DP

Takashi Sakamoto (1):
ALSA: oxfw: add an comment to Kconfig for TASCAM FireOne

---
sound/firewire/Kconfig | 1 +
sound/pci/Kconfig | 24 ++++++++++++------------
sound/pci/ctxfi/ctamixer.c | 6 +++---
sound/pci/ctxfi/ctamixer.h | 2 +-
sound/pci/ctxfi/ctdaio.c | 10 +++++-----
sound/pci/ctxfi/ctdaio.h | 4 ++--
sound/pci/ctxfi/ctresource.c | 2 +-
sound/pci/ctxfi/ctresource.h | 2 +-
sound/pci/ctxfi/ctsrc.c | 6 +++---
sound/pci/ctxfi/ctsrc.h | 4 ++--
sound/pci/hda/hda_jack.c | 18 ++----------------
sound/pci/hda/hda_jack.h | 2 +-
sound/pci/hda/patch_hdmi.c | 10 +++++++---
sound/sound_core.c | 2 +-
sound/usb/quirks.c | 1 +
15 files changed, 43 insertions(+), 51 deletions(-)

diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index bee0e5f1a116..e92a6d949847 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -38,6 +38,7 @@ config SND_OXFW
* Mackie(Loud) Tapco Link.Firewire
* Mackie(Loud) d.2 pro/d.4 pro
* Mackie(Loud) U.420/U.420d
+ * TASCAM FireOne

To compile this driver as a module, choose M here: the module
will be called snd-oxfw.
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index edfc1b8d553e..656ce39bddbc 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -25,7 +25,7 @@ config SND_ALS300
select SND_PCM
select SND_AC97_CODEC
select SND_OPL3_LIB
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+

@@ -50,7 +50,7 @@ config SND_ALI5451
tristate "ALi M5451 PCI Audio Controller"
select SND_MPU401_UART
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for the integrated AC97 sound
device on motherboards using the ALi M5451 Audio Controller
@@ -155,7 +155,7 @@ config SND_AZT3328
select SND_PCM
select SND_RAWMIDI
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for Aztech AZF3328 (PCI168)
soundcards.
@@ -463,7 +463,7 @@ config SND_EMU10K1
select SND_HWDEP
select SND_RAWMIDI
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y to include support for Sound Blaster PCI 512, Live!,
Audigy and E-mu APS (partially supported) soundcards.
@@ -479,7 +479,7 @@ config SND_EMU10K1X
tristate "Emu10k1X (Dell OEM Version)"
select SND_AC97_CODEC
select SND_RAWMIDI
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for the Dell OEM version of the
Sound Blaster Live!.
@@ -513,7 +513,7 @@ config SND_ES1938
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on ESS Solo-1
(ES1938, ES1946, ES1969) chips.
@@ -525,7 +525,7 @@ config SND_ES1968
tristate "ESS ES1968/1978 (Maestro-1/2/2E)"
select SND_MPU401_UART
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on ESS Maestro
1/2/2E chips.
@@ -612,7 +612,7 @@ config SND_ICE1712
select SND_MPU401_UART
select SND_AC97_CODEC
select BITREVERSE
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on the
ICE1712 (Envy24) chip.
@@ -700,7 +700,7 @@ config SND_LX6464ES
config SND_MAESTRO3
tristate "ESS Allegro/Maestro3"
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on ESS Maestro 3
(Allegro) chips.
@@ -806,7 +806,7 @@ config SND_SIS7019
tristate "SiS 7019 Audio Accelerator"
depends on X86_32
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for the SiS 7019 Audio Accelerator.

@@ -818,7 +818,7 @@ config SND_SONICVIBES
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on the S3
SonicVibes chip.
@@ -830,7 +830,7 @@ config SND_TRIDENT
tristate "Trident 4D-Wave DX/NX; SiS 7018"
select SND_MPU401_UART
select SND_AC97_CODEC
- select ZONE_DMA
+ depends on ZONE_DMA
help
Say Y here to include support for soundcards based on Trident
4D-Wave DX/NX or SiS 7018 chips.
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index c7dc38d41b7f..5fcbb065d870 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -49,7 +49,7 @@ static int amixer_output_slot(const struct rsc *rsc)
return (amixer_index(rsc) << 4) + 0x4;
}

-static struct rsc_ops amixer_basic_rsc_ops = {
+static const struct rsc_ops amixer_basic_rsc_ops = {
.master = amixer_master,
.next_conj = amixer_next_conj,
.index = amixer_index,
@@ -186,7 +186,7 @@ static int amixer_setup(struct amixer *amixer, struct rsc *input,
return 0;
}

-static struct amixer_rsc_ops amixer_ops = {
+static const struct amixer_rsc_ops amixer_ops = {
.set_input = amixer_set_input,
.set_invalid_squash = amixer_set_invalid_squash,
.set_scale = amixer_set_y,
@@ -357,7 +357,7 @@ static int sum_output_slot(const struct rsc *rsc)
return (sum_index(rsc) << 4) + 0xc;
}

-static struct rsc_ops sum_basic_rsc_ops = {
+static const struct rsc_ops sum_basic_rsc_ops = {
.master = sum_master,
.next_conj = sum_next_conj,
.index = sum_index,
diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h
index 72f42f27434e..2de18aa6508d 100644
--- a/sound/pci/ctxfi/ctamixer.h
+++ b/sound/pci/ctxfi/ctamixer.h
@@ -58,7 +58,7 @@ struct amixer {
unsigned char idx[8];
struct rsc *input; /* pointer to a resource acting as source */
struct sum *sum; /* Put amixer output to this summation node */
- struct amixer_rsc_ops *ops; /* AMixer specific operations */
+ const struct amixer_rsc_ops *ops; /* AMixer specific operations */
};

struct amixer_rsc_ops {
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 9b87dd28de83..7f089cb433e1 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -83,21 +83,21 @@ static int daio_in_next_conj_20k2(struct rsc *rsc)
return rsc->conj += 0x100;
}

-static struct rsc_ops daio_out_rsc_ops = {
+static const struct rsc_ops daio_out_rsc_ops = {
.master = daio_master,
.next_conj = daio_out_next_conj,
.index = daio_index,
.output_slot = NULL,
};

-static struct rsc_ops daio_in_rsc_ops_20k1 = {
+static const struct rsc_ops daio_in_rsc_ops_20k1 = {
.master = daio_master,
.next_conj = daio_in_next_conj_20k1,
.index = NULL,
.output_slot = daio_index,
};

-static struct rsc_ops daio_in_rsc_ops_20k2 = {
+static const struct rsc_ops daio_in_rsc_ops_20k2 = {
.master = daio_master,
.next_conj = daio_in_next_conj_20k2,
.index = NULL,
@@ -263,7 +263,7 @@ static int dao_clear_right_input(struct dao *dao)
return 0;
}

-static struct dao_rsc_ops dao_ops = {
+static const struct dao_rsc_ops dao_ops = {
.set_spos = dao_spdif_set_spos,
.commit_write = dao_commit_write,
.get_spos = dao_spdif_get_spos,
@@ -318,7 +318,7 @@ static int dai_commit_write(struct dai *dai)
return 0;
}

-static struct dai_rsc_ops dai_ops = {
+static const struct dai_rsc_ops dai_ops = {
.set_srt_srcl = dai_set_srt_srcl,
.set_srt_srcr = dai_set_srt_srcr,
.set_srt_msr = dai_set_srt_msr,
diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h
index 0ebbf350f51a..a30be73b08ea 100644
--- a/sound/pci/ctxfi/ctdaio.h
+++ b/sound/pci/ctxfi/ctdaio.h
@@ -51,7 +51,7 @@ struct daio {

struct dao {
struct daio daio;
- struct dao_rsc_ops *ops; /* DAO specific operations */
+ const struct dao_rsc_ops *ops; /* DAO specific operations */
struct imapper **imappers;
struct daio_mgr *mgr;
struct hw *hw;
@@ -60,7 +60,7 @@ struct dao {

struct dai {
struct daio daio;
- struct dai_rsc_ops *ops; /* DAI specific operations */
+ const struct dai_rsc_ops *ops; /* DAI specific operations */
struct hw *hw;
void *ctrl_blk;
};
diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c
index 1a97e406d8ec..c5124c3c0fd1 100644
--- a/sound/pci/ctxfi/ctresource.c
+++ b/sound/pci/ctxfi/ctresource.c
@@ -127,7 +127,7 @@ static int rsc_master(struct rsc *rsc)
return rsc->conj = rsc->idx;
}

-static struct rsc_ops rsc_generic_ops = {
+static const struct rsc_ops rsc_generic_ops = {
.index = rsc_index,
.output_slot = audio_ring_slot,
.master = rsc_master,
diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h
index 9b746c3719e6..736d9f7e9e16 100644
--- a/sound/pci/ctxfi/ctresource.h
+++ b/sound/pci/ctxfi/ctresource.h
@@ -39,7 +39,7 @@ struct rsc {
u32 msr:4; /* The Master Sample Rate a resource working on */
void *ctrl_blk; /* Chip specific control info block for a resource */
struct hw *hw; /* Chip specific object for hardware access means */
- struct rsc_ops *ops; /* Generic resource operations */
+ const struct rsc_ops *ops; /* Generic resource operations */
};

struct rsc_ops {
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index ec1f08464d93..a5a72df29801 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -335,7 +335,7 @@ static int src_default_config_arcrw(struct src *src)
return 0;
}

-static struct src_rsc_ops src_rsc_ops = {
+static const struct src_rsc_ops src_rsc_ops = {
.set_state = src_set_state,
.set_bm = src_set_bm,
.set_sf = src_set_sf,
@@ -611,7 +611,7 @@ static int srcimp_index(const struct rsc *rsc)
return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj];
}

-static struct rsc_ops srcimp_basic_rsc_ops = {
+static const struct rsc_ops srcimp_basic_rsc_ops = {
.master = srcimp_master,
.next_conj = srcimp_next_conj,
.index = srcimp_index,
@@ -662,7 +662,7 @@ static int srcimp_unmap(struct srcimp *srcimp)
return 0;
}

-static struct srcimp_rsc_ops srcimp_ops = {
+static const struct srcimp_rsc_ops srcimp_ops = {
.map = srcimp_map,
.unmap = srcimp_unmap
};
diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h
index da7573c5db9b..92944a012bc4 100644
--- a/sound/pci/ctxfi/ctsrc.h
+++ b/sound/pci/ctxfi/ctsrc.h
@@ -48,7 +48,7 @@ struct src_rsc_ops;
struct src {
struct rsc rsc; /* Basic resource info */
struct src *intlv; /* Pointer to next interleaved SRC in a series */
- struct src_rsc_ops *ops; /* SRC specific operations */
+ const struct src_rsc_ops *ops; /* SRC specific operations */
/* Number of contiguous srcs for interleaved usage */
unsigned char multi;
unsigned char mode; /* Working mode of this SRC resource */
@@ -110,7 +110,7 @@ struct srcimp {
struct imapper *imappers;
unsigned int mapped; /* A bit-map indicating which conj rsc is mapped */
struct srcimp_mgr *mgr;
- struct srcimp_rsc_ops *ops;
+ const struct srcimp_rsc_ops *ops;
};

struct srcimp_rsc_ops {
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 366efbf87d41..c945e257d368 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -383,7 +383,7 @@ static void hda_free_jack_priv(struct snd_jack *jack)
* This assigns a jack-detection kctl to the given pin. The kcontrol
* will have the given name and index.
*/
-static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
+int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
const char *name, bool phantom_jack)
{
struct hda_jack_tbl *jack;
@@ -410,20 +410,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,

return 0;
}
-
-/**
- * snd_hda_jack_add_kctl - Add a jack kctl for the given pin
- * @codec: the HDA codec
- * @nid: pin NID
- * @name: the name string for the jack ctl
- *
- * This is a simple helper calling __snd_hda_jack_add_kctl().
- */
-int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
- const char *name)
-{
- return __snd_hda_jack_add_kctl(codec, nid, name, false);
-}
EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl);

static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
@@ -451,7 +437,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
if (phantom_jack)
/* Example final name: "Internal Mic Phantom Jack" */
strncat(name, " Phantom", sizeof(name) - strlen(name) - 1);
- err = __snd_hda_jack_add_kctl(codec, nid, name, phantom_jack);
+ err = snd_hda_jack_add_kctl(codec, nid, name, phantom_jack);
if (err < 0)
return err;

diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 387d30984dfe..858708a044f5 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -82,7 +82,7 @@ static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);

int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
- const char *name);
+ const char *name, bool phantom_jack);
int snd_hda_jack_add_kctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg);

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index f503a883bef3..60cd9e700909 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -50,8 +50,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
#define is_haswell(codec) ((codec)->core.vendor_id == 0x80862807)
#define is_broadwell(codec) ((codec)->core.vendor_id == 0x80862808)
#define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809)
+#define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a)
#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \
- || is_skylake(codec))
+ || is_skylake(codec) || is_broxton(codec))

#define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882)
#define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883)
@@ -2096,14 +2097,17 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
struct hdmi_spec *spec = codec->spec;
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
int pcmdev = get_pcm_rec(spec, pin_idx)->device;
+ bool phantom_jack;

if (pcmdev > 0)
sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
- if (!is_jack_detectable(codec, per_pin->pin_nid))
+ phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid);
+ if (phantom_jack)
strncat(hdmi_str, " Phantom",
sizeof(hdmi_str) - strlen(hdmi_str) - 1);

- return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str);
+ return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str,
+ phantom_jack);
}

static int generic_hdmi_build_controls(struct hda_codec *codec)
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 11e953a1fa45..99b73c675743 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -655,7 +655,7 @@ static void cleanup_oss_soundcore(void)
static int __init init_oss_soundcore(void)
{
if (preclaim_oss &&
- register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) == -1) {
+ register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) < 0) {
printk(KERN_ERR "soundcore: sound device already in use.\n");
return -EBUSY;
}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 4897ea171194..5ca80e7d30cd 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1274,6 +1274,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */
case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */
case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */
+ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */
if (fp->altsetting == 3)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
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