From cb68429d155507ec46f1922d2beb7bc637de4836 Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 7 Nov 2015 01:56:05 +0300 Subject: sound: fix check for error condition of register_chrdev() init_oss_soundcore() compares returned value of register_chrdev() with -1, while other error codes can be returned. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Signed-off-by: Takashi Iwai diff --git a/sound/sound_core.c b/sound/sound_core.c index 11e953a..99b73c6 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; } -- cgit v0.10.2 From 94d505e9746a53af79094904ff0aa0cf0952ed58 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 8 Nov 2015 14:46:34 +0900 Subject: ALSA: oxfw: add an comment to Kconfig for TASCAM FireOne A commit to add support for this model should have added a comment about this model to Kconfig. Fixes: 759a2f40c9fa('ALSA: oxfw: add an entry for TASCAM FireOne') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index bee0e5f..e92a6d9 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. -- cgit v0.10.2 From 16771c7c704769c5f3d70c024630b6e5b3eafa67 Mon Sep 17 00:00:00 2001 From: Jurgen Kramer Date: Mon, 9 Nov 2015 12:13:55 +0100 Subject: ALSA: usb: Add native DSD support for Aune X1S This patch adds native DSD support for the Aune X1S 32BIT/384 DSD DAC Signed-off-by: Jurgen Kramer Cc: Signed-off-by: Takashi Iwai diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4897ea1..5ca80e7 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; -- cgit v0.10.2 From 43f2cdeb7a61598050028f1eae51b9cb5398af42 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 8 Nov 2015 23:40:41 +0100 Subject: ALSA: ctxfi: constify rsc ops structures The various rsc ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index c7dc38d..5fcbb06 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 72f42f2..2de18aa 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 9b87dd2..7f089cb 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 0ebbf35..a30be73 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 1a97e40..c5124c3 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 9b746c3..736d9f7 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 ec1f084..a5a72df 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 da7573c..92944a0 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 { -- cgit v0.10.2 From e2656412f2a7343ecfd13eb74bac0a6e6e9c5aad Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Wed, 11 Nov 2015 16:54:27 +0800 Subject: ALSA: hda/hdmi - apply Skylake fix-ups to Broxton display codec Broxton and Skylake have the same behavior on display audio. So this patch applys Skylake fix-ups to Broxton. Signed-off-by: Lu, Han Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f503a88..309274b 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) -- cgit v0.10.2 From 909cadc6c8c7e52149fb4687453277d6cabe8c80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Nov 2015 11:52:13 +0100 Subject: ALSA: hda - Simplify phantom jack handling for HDMI/DP The HDMI codec parser may create a phantom jack, but the helper function snd_hda_jack_add_kctl() treats always as a normal jack. This is superfluous as the jack query is executed at each time the jack sync is performed. Since the HDMI codec parser is the only caller of this function, it's easier to change back this directly calling the original __snd_hda_jack_add_kctl() with phantom_jack parameter. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 366efbf..c945e25 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 387d309..858708a 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 309274b..60cd9e7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2097,14 +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) -- cgit v0.10.2 From 2db1a57986d37653583e67ccbf13082aadc8f25d Mon Sep 17 00:00:00 2001 From: Dan Williams Date: Thu, 12 Nov 2015 12:13:57 -0800 Subject: ALSA: pci: depend on ZONE_DMA There are several sound drivers that 'select ZONE_DMA'. This is backwards as ZONE_DMA is an architecture capability exported to drivers. Switch the polarity of the dependency to disable these drivers when the architecture does not support ZONE_DMA. This was discovered in the context of testing/enabling devm_memremap_pages() which depends on ZONE_DEVICE. ZONE_DEVICE in turn depends on !ZONE_DMA. Reported-by: Jeff Moyer Signed-off-by: Dan Williams Signed-off-by: Takashi Iwai diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index edfc1b8..656ce39 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. -- cgit v0.10.2