diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-12-20 03:07:17 +0100 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-12-20 03:07:17 +0100 |
commit | 20e471fd34d1f79bed65fdc1bf4ad090f70472a5 (patch) | |
tree | 7e74e4ca71a90c6b39392dacab07e26feedb8185 /sound | |
parent | Merge branch 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/nab/... (diff) | |
parent | ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC (diff) | |
download | linux-20e471fd34d1f79bed65fdc1bf4ad090f70472a5.tar.xz linux-20e471fd34d1f79bed65fdc1bf4ad090f70472a5.zip |
Merge tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few fixes that have landed after the previous pull request.
All are driver specific fixes including:
- error/int value fixes in OXFW,
- Intel Skylake HD-audio HDMI codec support,
- Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
- a few more DSD support and a quirk for Arcam rPAC in usb-audio,
- a typo fix for Scarlett 6i6,
- fixes for new ASIHPI firmware,
- ASoC Exynos7 cleanups,
- Intel ACPI support, and
- a fix for PCM512 register cache sync"
* tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
ALSA: hda/realtek - New codec support for ALC298
ALSA: asihpi: update to HPI version 4.14
ALSA: asihpi: increase tuner pad cache size
ALSA: asihpi: relax firmware version check
ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
ALSA: hda - Add quirk for Packard Bell EasyNote MX65
ALSA: usb-audio: add native DSD support for Matrix Audio DACs
ALSA: hda/realtek - New codec support for ALC256
ALSA: hda/realtek - Add new Dell desktop for ALC3234 headset mode
ASoC: Intel: fix possible acpi enumeration panic
ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
ASoC: Intel: fix return value check in sst_acpi_probe()
ALSA: hda - Make add_stereo_mix_input flag tristate
ALSA: hda - Create capture source ctls when stereo mix input is added
ALSA: hda - Fix typos in snd_hda_get_int_hint() kerneldoc comments
ALSA: hda - add codec ID for Skylake display audio codec
ALSA: oxfw: some signedness bugs
ALSA: oxfw: fix detect_loud_models() return value
ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
...
Diffstat (limited to 'sound')
-rw-r--r-- | sound/firewire/oxfw/oxfw-pcm.c | 6 | ||||
-rw-r--r-- | sound/firewire/oxfw/oxfw-proc.c | 2 | ||||
-rw-r--r-- | sound/firewire/oxfw/oxfw-stream.c | 3 | ||||
-rw-r--r-- | sound/firewire/oxfw/oxfw.c | 2 | ||||
-rw-r--r-- | sound/pci/asihpi/hpi_internal.h | 6 | ||||
-rw-r--r-- | sound/pci/asihpi/hpi_version.h | 6 | ||||
-rw-r--r-- | sound/pci/asihpi/hpidspcd.c | 26 | ||||
-rw-r--r-- | sound/pci/hda/hda_generic.c | 10 | ||||
-rw-r--r-- | sound/pci/hda/hda_generic.h | 9 | ||||
-rw-r--r-- | sound/pci/hda/hda_sysfs.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_analog.c | 42 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/patch_hdmi.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 15 | ||||
-rw-r--r-- | sound/pci/hda/patch_via.c | 2 | ||||
-rw-r--r-- | sound/soc/atmel/atmel_ssc_dai.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/Kconfig | 2 | ||||
-rw-r--r-- | sound/soc/codecs/pcm512x-i2c.c | 7 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.c | 4 | ||||
-rw-r--r-- | sound/soc/intel/sst/sst_acpi.c | 10 | ||||
-rw-r--r-- | sound/soc/samsung/i2s.c | 2 | ||||
-rw-r--r-- | sound/usb/mixer_maps.c | 15 | ||||
-rw-r--r-- | sound/usb/mixer_scarlett.c | 2 | ||||
-rw-r--r-- | sound/usb/quirks.c | 5 |
24 files changed, 139 insertions, 51 deletions
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9bc556b15a92..67ade0775a5b 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -19,7 +19,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, .min = UINT_MAX, .max = 0, .integer = 1 }; struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { if (formats[i] == NULL) @@ -47,7 +47,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_oxfw_stream_formation formation; - unsigned int i, j, err; + int i, j, err; unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0}; count = 0; @@ -80,7 +80,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) { struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; hw->channels_min = UINT_MAX; hw->channels_max = 0; diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 604808e5526d..8ba4f9f262b8 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -15,7 +15,7 @@ static void proc_read_formation(struct snd_info_entry *entry, struct snd_oxfw_stream_formation formation, curr; u8 *format; char flag; - unsigned int i, err; + int i, err; /* Show input. */ err = snd_oxfw_stream_get_current_formation(oxfw, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index b77cf80f1678..bda845afb470 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -61,7 +61,8 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, u8 **formats; struct snd_oxfw_stream_formation formation; enum avc_general_plug_dir dir; - unsigned int i, err, len; + unsigned int len; + int i, err; if (s == &oxfw->tx_stream) { formats = oxfw->tx_stream_formats; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cf1d0b55e827..60e5cad0531a 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -43,7 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit) err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); if (err < 0) - return err; + return false; for (i = 0; i < ARRAY_SIZE(models); i++) { if (strcmp(models[i], model) == 0) diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 48380ce2c81b..aeea679b2281 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1367,9 +1367,9 @@ struct hpi_control_cache_single { struct hpi_control_cache_pad { struct hpi_control_cache_info i; u32 field_valid_flags; - u8 c_channel[8]; - u8 c_artist[40]; - u8 c_title[40]; + u8 c_channel[40]; + u8 c_artist[100]; + u8 c_title[100]; u8 c_comment[200]; u32 pTY; u32 pI; diff --git a/sound/pci/asihpi/hpi_version.h b/sound/pci/asihpi/hpi_version.h index e9146e53bd50..6623ab110038 100644 --- a/sound/pci/asihpi/hpi_version.h +++ b/sound/pci/asihpi/hpi_version.h @@ -11,13 +11,13 @@ Production releases have even minor version. /* Use single digits for versions less that 10 to avoid octal. */ /* *** HPI_VER is the only edit required to update version *** */ /** HPI version */ -#define HPI_VER HPI_VERSION_CONSTRUCTOR(4, 10, 1) +#define HPI_VER HPI_VERSION_CONSTRUCTOR(4, 14, 3) /** HPI version string in dotted decimal format */ -#define HPI_VER_STRING "4.10.01" +#define HPI_VER_STRING "4.14.03" /** Library version as documented in hpi-api-versions.txt */ -#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 2, 0) +#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 4, 0) /** Construct hpi version number from major, minor, release numbers */ #define HPI_VERSION_CONSTRUCTOR(maj, min, r) ((maj << 16) + (min << 8) + r) diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index ac9163770013..3603c24f34d2 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -1,8 +1,9 @@ -/***********************************************************************/ -/** +/*********************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Functions for reading DSP code using hotplug firmware loader + + Copyright (C) 1997-2014 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -17,11 +18,7 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -\file -Functions for reading DSP code using -hotplug firmware loader from individual dsp code files -*/ -/***********************************************************************/ +***********************************************************************/ #define SOURCEFILE_NAME "hpidspcd.c" #include "hpidspcd.h" #include "hpidebug.h" @@ -68,17 +65,18 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, goto error2; } - if ((header.version >> 9) != (HPI_VER >> 9)) { - /* Consider even and subsequent odd minor versions to be compatible */ - dev_err(&dev->dev, "Incompatible firmware version DSP image %X != Driver %X\n", + if (HPI_VER_MAJOR(header.version) != HPI_VER_MAJOR(HPI_VER)) { + /* Major version change probably means Host-DSP protocol change */ + dev_err(&dev->dev, + "Incompatible firmware version DSP image %X != Driver %X\n", header.version, HPI_VER); goto error2; } if (header.version != HPI_VER) { - dev_info(&dev->dev, - "Firmware: release version mismatch DSP image %X != Driver %X\n", - header.version, HPI_VER); + dev_warn(&dev->dev, + "Firmware version mismatch: DSP image %X != Driver %X\n", + header.version, HPI_VER); } HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 63b69f750d8e..b680b4ec6331 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3218,12 +3218,13 @@ static int create_input_ctls(struct hda_codec *codec) } /* add stereo mix when explicitly enabled via hint */ - if (mixer && spec->add_stereo_mix_input && - snd_hda_get_bool_hint(codec, "add_stereo_mix_input") > 0) { + if (mixer && spec->add_stereo_mix_input == HDA_HINT_STEREO_MIX_ENABLE) { err = parse_capture_source(codec, mixer, CFG_IDX_MIX, num_adcs, "Stereo Mix", 0); if (err < 0) return err; + else + spec->suppress_auto_mic = 1; } return 0; @@ -4542,9 +4543,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, /* add stereo mix if available and not enabled yet */ if (!spec->auto_mic && spec->mixer_nid && - spec->add_stereo_mix_input && - spec->input_mux.num_items > 1 && - snd_hda_get_bool_hint(codec, "add_stereo_mix_input") < 0) { + spec->add_stereo_mix_input == HDA_HINT_STEREO_MIX_AUTO && + spec->input_mux.num_items > 1) { err = parse_capture_source(codec, spec->mixer_nid, CFG_IDX_MIX, spec->num_all_adcs, "Stereo Mix", 0); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 61dd5153f512..3d852660443a 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -222,7 +222,7 @@ struct hda_gen_spec { unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */ unsigned int indep_hp:1; /* independent HP supported */ unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ - unsigned int add_stereo_mix_input:1; /* add aamix as a capture src */ + unsigned int add_stereo_mix_input:2; /* add aamix as a capture src */ unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */ unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */ @@ -291,6 +291,13 @@ struct hda_gen_spec { struct hda_jack_callback *cb); }; +/* values for add_stereo_mix_input flag */ +enum { + HDA_HINT_STEREO_MIX_DISABLE, /* No stereo mix input */ + HDA_HINT_STEREO_MIX_ENABLE, /* Add stereo mix input */ + HDA_HINT_STEREO_MIX_AUTO, /* Add only if auto-mic is disabled */ +}; + int snd_hda_gen_spec_init(struct hda_gen_spec *spec); int snd_hda_gen_init(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index bef721592c3a..ccc962a1699f 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -468,7 +468,7 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) EXPORT_SYMBOL_GPL(snd_hda_get_bool_hint); /** - * snd_hda_get_bool_hint - Get a boolean hint value + * snd_hda_get_int_hint - Get an integer hint value * @codec: the HDA codec * @key: the hint key string * @valp: pointer to store a value diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c81b715d6c98..a9d78e275138 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -195,7 +195,8 @@ static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp) codec->no_sticky_stream = 1; spec->gen.indep_hp = indep_hp; - spec->gen.add_stereo_mix_input = 1; + if (!spec->gen.add_stereo_mix_input) + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); if (err < 0) @@ -256,6 +257,18 @@ static void ad1986a_fixup_eapd(struct hda_codec *codec, } } +/* enable stereo-mix input for avoiding regression on KDE (bko#88251) */ +static void ad1986a_fixup_eapd_mix_in(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + ad1986a_fixup_eapd(codec, fix, action); + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_ENABLE; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -264,6 +277,8 @@ enum { AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, AD1986A_FIXUP_EAPD, + AD1986A_FIXUP_EAPD_MIX_IN, + AD1986A_FIXUP_EASYNOTE, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -328,6 +343,30 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad1986a_fixup_eapd, }, + [AD1986A_FIXUP_EAPD_MIX_IN] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd_mix_in, + }, + [AD1986A_FIXUP_EASYNOTE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0421402f }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x90a70130 }, /* int mic */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x04a19040 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x21, 0x411111f0 }, /* N/A */ + { 0x22, 0x411111f0 }, /* N/A */ + { 0x23, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + { 0x25, 0x411111f0 }, /* N/A */ + {} + }, + .chained = true, + .chain_id = AD1986A_FIXUP_EAPD_MIX_IN, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -341,6 +380,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), + SND_PCI_QUIRK(0x1631, 0xc022, "PackardBell EasyNote MX65", AD1986A_FIXUP_EASYNOTE), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e9ebc7bd752c..fd3ed18670e9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -855,14 +855,14 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15045: codec->single_adc_amp = 1; spec->gen.mixer_nid = 0x17; - spec->gen.add_stereo_mix_input = 1; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; snd_hda_pick_fixup(codec, cxt5045_fixup_models, cxt5045_fixups, cxt_fixups); break; case 0x14f15047: codec->pin_amp_workaround = 1; spec->gen.mixer_nid = 0x19; - spec->gen.add_stereo_mix_input = 1; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; snd_hda_pick_fixup(codec, cxt5047_fixup_models, cxt5047_fixups, cxt_fixups); break; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9dc9cf8c90e9..5f13d2d18079 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -47,7 +47,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) #define is_broadwell(codec) ((codec)->vendor_id == 0x80862808) -#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) +#define is_skylake(codec) ((codec)->vendor_id == 0x80862809) +#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ + || is_skylake(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) @@ -3365,6 +3367,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, @@ -3425,6 +3428,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862808"); +MODULE_ALIAS("snd-hda-codec-id:80862809"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:80862883"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a722067c491c..65f1f4e18ea5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -321,10 +321,12 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) break; case 0x10ec0233: case 0x10ec0255: + case 0x10ec0256: case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: + case 0x10ec0298: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; case 0x10ec0285: @@ -2659,7 +2661,9 @@ enum { ALC269_TYPE_ALC284, ALC269_TYPE_ALC285, ALC269_TYPE_ALC286, + ALC269_TYPE_ALC298, ALC269_TYPE_ALC255, + ALC269_TYPE_ALC256, }; /* @@ -2686,7 +2690,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC282: case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: + case ALC269_TYPE_ALC298: case ALC269_TYPE_ALC255: + case ALC269_TYPE_ALC256: ssids = alc269_ssids; break; default: @@ -4829,6 +4835,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -5417,9 +5424,15 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC286; spec->shutup = alc286_shutup; break; + case 0x10ec0298: + spec->codec_variant = ALC269_TYPE_ALC298; + break; case 0x10ec0255: spec->codec_variant = ALC269_TYPE_ALC255; break; + case 0x10ec0256: + spec->codec_variant = ALC269_TYPE_ALC256; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -6341,6 +6354,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, + { .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, @@ -6360,6 +6374,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, { .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 }, + { .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6c206b6c8d65..3de6d3d779c9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -137,7 +137,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; - spec->gen.add_stereo_mix_input = 1; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; return spec; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index b1cc2a4a7fc0..99ff35e2a25d 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, if (!ssc_p->dir_mask) { if (ssc_p->initialized) { /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); + pr_debug("atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); free_irq(ssc_p->ssc->irq, ssc_p); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883c5778b309..8349f982a586 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -520,6 +520,8 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + select REGMAP_I2C + select REGMAP_IRQ config SND_SOC_RT5677_SPI tristate diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 4d62230bd378..d0547fa275fc 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -24,8 +24,13 @@ static int pcm512x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct regmap *regmap; + struct regmap_config config = pcm512x_regmap; - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + /* msb needs to be set to enable auto-increment of addresses */ + config.read_flag_mask = 0x80; + config.write_flag_mask = 0x80; + + regmap = devm_regmap_init_i2c(i2c, &config); if (IS_ERR(regmap)) return PTR_ERR(regmap); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7789a8726e3..27141e2df878 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2209,6 +2209,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) int gpio_state, jack_type = 0; unsigned int val; + if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + dev_err(codec->dev, "invalid gpio\n"); + return -EINVAL; + } gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa4434e..3abc29e8a928 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -43,7 +43,7 @@ #include "sst.h" struct sst_machines { - char codec_id[32]; + char *codec_id; char board[32]; char machine[32]; void (*machine_quirk)(void); @@ -277,16 +277,16 @@ int sst_acpi_probe(struct platform_device *pdev) dev_dbg(dev, "ACPI device id: %x\n", dev_id); plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); - if (plat_dev == NULL) { + if (IS_ERR(plat_dev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); - return -ENODEV; + return PTR_ERR(plat_dev); } /* Create platform device for sst machine driver */ mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); - if (mdev == NULL) { + if (IS_ERR(mdev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->machine); - return -ENODEV; + return PTR_ERR(mdev); } ret = sst_alloc_drv_context(&ctx, dev, dev_id); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b1a7c5bce4a1..b5a80c528d86 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1261,6 +1261,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } + + sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1994d41348f8..b703cb3cda19 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -333,8 +333,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; -static const struct usbmix_name_map kef_x300a_map[] = { - { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ +/* some (all?) SCMS USB3318 devices are affected by a firmware lock up + * when anything attempts to access FU 10 (control) + */ +static const struct usbmix_name_map scms_usb3318_map[] = { + { 10, NULL }, { 0 } }; @@ -434,8 +437,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = ebox44_map, }, { + /* KEF X300A */ .id = USB_ID(0x27ac, 0x1000), - .map = kef_x300a_map, + .map = scms_usb3318_map, + }, + { + /* Arcam rPAC */ + .id = USB_ID(0x25c4, 0x0003), + .map = scms_usb3318_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 9109652b88b9..7438e7c4a842 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -655,7 +655,7 @@ static struct scarlett_device_info s6i6_info = { .names = NULL }, - .num_controls = 0, + .num_controls = 9, .controls = { { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4dbfb3d18ee2..a7398412310b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1245,8 +1245,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - /* iFi Audio micro/nano iDSD */ - case USB_ID(0x20b1, 0x3008): + case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ + case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ + case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; |