diff options
author | Arnd Bergmann <arnd@arndb.de> | 2012-07-21 23:34:31 +0200 |
---|---|---|
committer | Arnd Bergmann <arnd@arndb.de> | 2012-07-21 23:34:31 +0200 |
commit | 088eb2a84b3b093af60bbe0b65151e8a9cd2d57d (patch) | |
tree | 8f145d8c71f6e249d6664b5482ac216b8c6e4016 /sound | |
parent | 6217a69a1e38807b3046b4fd6e4bf413e0e03b67 (diff) | |
parent | 84a1caf1453c3d44050bd22db958af4a7f99315c (diff) |
Merge tag 'v3.5-rc7' into late/soc
Linux 3.5-rc7
Diffstat (limited to 'sound')
-rw-r--r-- | sound/i2c/other/tea575x-tuner.c | 21 | ||||
-rw-r--r-- | sound/pci/es1968.c | 2 | ||||
-rw-r--r-- | sound/pci/fm801.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/Kconfig | 13 | ||||
-rw-r--r-- | sound/pci/hda/hda_auto_parser.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_auto_parser.h | 10 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 50 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 43 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 7 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.h | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm2200.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8904.c | 26 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm8996.c | 8 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 38 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_wm8903.c | 13 | ||||
-rw-r--r-- | sound/usb/6fire/firmware.c | 2 | ||||
-rw-r--r-- | sound/usb/endpoint.c | 73 | ||||
-rw-r--r-- | sound/usb/mixer_maps.c | 8 | ||||
-rw-r--r-- | sound/usb/pcm.c | 82 | ||||
-rw-r--r-- | sound/usb/quirks-table.h | 30 |
25 files changed, 256 insertions, 194 deletions
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 582aace20ea3..7eca25fae413 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -37,8 +37,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); -#define FREQ_LO (76U * 16000) -#define FREQ_HI (108U * 16000) +#define FREQ_LO ((tea->tea5759 ? 760 : 875) * 1600U) +#define FREQ_HI ((tea->tea5759 ? 910 : 1080) * 1600U) /* * definitions @@ -120,9 +120,9 @@ static u32 snd_tea575x_read(struct snd_tea575x *tea) return data; } -static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) +static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) { - u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; + u32 freq = val & TEA575X_BIT_FREQ_MASK; if (freq == 0) return freq; @@ -139,6 +139,11 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ } +static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq; @@ -156,6 +161,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= freq & TEA575X_BIT_FREQ_MASK; snd_tea575x_write(tea, tea->val); + tea->freq = snd_tea575x_val_to_freq(tea, tea->val); } /* @@ -317,7 +323,6 @@ static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) } static const struct v4l2_file_operations tea575x_fops = { - .owner = THIS_MODULE, .unlocked_ioctl = video_ioctl2, .open = v4l2_fh_open, .release = v4l2_fh_release, @@ -337,7 +342,6 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { }; static const struct video_device tea575x_radio = { - .fops = &tea575x_fops, .ioctl_ops = &tea575x_ioctl_ops, .release = video_device_release_empty, }; @@ -349,7 +353,7 @@ static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { /* * initialize all the tea575x chips */ -int snd_tea575x_init(struct snd_tea575x *tea) +int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner) { int retval; @@ -374,6 +378,9 @@ int snd_tea575x_init(struct snd_tea575x *tea) tea->vd.lock = &tea->mutex; tea->vd.v4l2_dev = tea->v4l2_dev; tea->vd.ctrl_handler = &tea->ctrl_handler; + tea->fops = tea575x_fops; + tea->fops.owner = owner; + tea->vd.fops = &tea->fops; set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags); /* disable hw_freq_seek if we can't use it */ if (tea->cannot_read_data) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 67f47d891959..52b5c0bf90c1 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2769,7 +2769,7 @@ static int __devinit snd_es1968_create(struct snd_card *card, chip->tea.ops = &snd_es1968_tea_ops; strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea)) + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) printk(KERN_INFO "es1968: detected TEA575x radio\n"); #endif diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index f69662322750..b32e8024ea86 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1254,7 +1254,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - if (snd_tea575x_init(&chip->tea)) { + if (snd_tea575x_init(&chip->tea, THIS_MODULE)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); snd_fm801_free(chip); return -ENODEV; @@ -1263,7 +1263,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, /* autodetect tuner connection */ for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { chip->tea575x_tuner = tea575x_tuner; - if (!snd_tea575x_init(&chip->tea)) { + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { snd_printk(KERN_INFO "detected TEA575x radio type %s\n", get_tea575x_gpio(chip)->name); break; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 163b6b5de3eb..d03079764189 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -97,19 +97,6 @@ config SND_HDA_CODEC_REALTEK snd-hda-codec-realtek. This module is automatically loaded at probing. -config SND_HDA_ENABLE_REALTEK_QUIRKS - bool "Build static quirks for Realtek codecs" - depends on SND_HDA_CODEC_REALTEK - default y - help - Say Y here to build the static quirks codes for Realtek codecs. - If you need the "model" preset that the default BIOS auto-parser - can't handle, turn this option on. - - If your device works with model=auto option, basically you don't - need the quirk code. By turning this off, you can reduce the - module size quite a lot. - config SND_HDA_CODEC_ANALOG bool "Build Analog Device HD-audio codec support" default y diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 6e9ef3e25093..f7520b9f909c 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -618,7 +618,6 @@ int snd_hda_gen_add_verbs(struct hda_gen_spec *spec, const struct hda_verb *list) { const struct hda_verb **v; - snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8); v = snd_array_new(&spec->verbs); if (!v) return -ENOMEM; diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index 2a7889dfbd1b..632ad0ad3007 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -157,4 +157,14 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct hda_fixup *fixlist); +static inline void snd_hda_gen_init(struct hda_gen_spec *spec) +{ + snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8); +} + +static inline void snd_hda_gen_free(struct hda_gen_spec *spec) +{ + snd_array_free(&spec->verbs); +} + #endif /* __SOUND_HDA_AUTO_PARSER_H */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41ca803a1fff..51cb2a2e4fce 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1184,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + snd_hda_jack_tbl_clear(codec); restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); @@ -1192,6 +1193,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) list_del(&codec->list); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); + snd_array_free(&codec->cvt_setups); snd_array_free(&codec->conn_lists); snd_array_free(&codec->spdif_out); codec->bus->caddr_tbl[codec->addr] = NULL; @@ -2333,6 +2335,8 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* free only driver_pins so that init_pins + user_pins are restored */ snd_array_free(&codec->driver_pins); restore_pincfgs(codec); + snd_array_free(&codec->cvt_setups); + snd_array_free(&codec->spdif_out); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; @@ -4393,20 +4397,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition > 0) { + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { spin_unlock(&codec->power_lock); return; } @@ -4430,8 +4433,37 @@ void snd_hda_power_up(struct hda_codec *codec) codec->power_transition = 0; spin_unlock(&codec->power_lock); } + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); +} EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960c8591..2fdaadbb4326 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 02763827dde0..7757536b9d5f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b3793b..2bf99fc1cbf2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -445,8 +445,10 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + snd_hda_gen_free(&spec->gen); snd_hda_detach_beep_device(codec); - kfree(codec->spec); + kfree(spec); } static const struct snd_kcontrol_new cxt_capture_mixers[] = { @@ -4466,6 +4468,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4497,6 +4500,7 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); switch (codec->vendor_id) { case 0x14f15045: diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f8f4906e498d..aa4c25e0f327 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2289,6 +2289,7 @@ static void alc_free(struct hda_codec *codec) alc_shutup(codec); alc_free_kctls(codec); alc_free_bind_ctls(codec); + snd_hda_gen_free(&spec->gen); kfree(spec); snd_hda_detach_beep_device(codec); } @@ -4253,6 +4254,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) return -ENOMEM; codec->spec = spec; spec->mixer_nid = mixer_nid; + snd_hda_gen_init(&spec->gen); err = alc_codec_rename_from_preset(codec); if (err < 0) { @@ -6686,6 +6688,31 @@ static const struct alc_model_fixup alc662_fixup_models[] = { {} }; +static void alc662_fill_coef(struct hda_codec *codec) +{ + int val, coef; + + coef = alc_get_coef0(codec); + + switch (codec->vendor_id) { + case 0x10ec0662: + if ((coef & 0x00f0) == 0x0030) { + val = alc_read_coef_idx(codec, 0x4); /* EAPD Ctrl */ + alc_write_coef_idx(codec, 0x4, val & ~(1<<10)); + } + break; + case 0x10ec0272: + case 0x10ec0273: + case 0x10ec0663: + case 0x10ec0665: + case 0x10ec0670: + case 0x10ec0671: + case 0x10ec0672: + val = alc_read_coef_idx(codec, 0xd); /* EAPD Ctrl */ + alc_write_coef_idx(codec, 0xd, val | (1<<14)); + break; + } +} /* */ @@ -6705,12 +6732,8 @@ static int patch_alc662(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); - if ((alc_get_coef0(codec) & (1 << 14)) && - codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) { - if (alc_codec_rename(codec, "ALC272X") < 0) - goto error; - } + spec->init_hook = alc662_fill_coef; + alc662_fill_coef(codec); alc_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); @@ -6718,6 +6741,13 @@ static int patch_alc662(struct hda_codec *codec) alc_auto_parse_customize_define(codec); + if ((alc_get_coef0(codec) & (1 << 14)) && + codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) { + if (alc_codec_rename(codec, "ALC272X") < 0) + goto error; + } + /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) @@ -6800,6 +6830,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, + { .id = 0x10ec0280, .name = "ALC280", .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_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7db8228f1b88..07675282015a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4367,7 +4367,7 @@ static int stac92xx_init(struct hda_codec *codec) AC_PINCTL_IN_EN); for (i = 0; i < spec->num_pwrs; i++) { hda_nid_t nid = spec->pwr_nids[i]; - int pinctl, def_conf; + unsigned int pinctl, def_conf; def_conf = snd_hda_codec_get_pincfg(codec, nid); def_conf = get_defcfg_connect(def_conf); @@ -4376,6 +4376,11 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 0); continue; } + if (def_conf == AC_JACK_PORT_FIXED) { + /* no need for jack detection for fixed pins */ + stac_toggle_power_map(codec, nid, 1); + continue; + } /* power on when no jack detection is available */ /* or when the VREF is used for controlling LED */ if (!spec->hp_detect || diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4fa34b2..e9b62b5ea637 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb60683..08c7f6685ff0 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index acbdc5fde923..32682c1b7cde 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = { static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { 5644800, + 3763200, 2882400, 1881600, 1411200, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d74c54..812acd83fb48 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } + regcache_cache_only(wm8904->regmap, false); regcache_sync(wm8904->regmap); /* Enable bias */ @@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); -#ifdef CONFIG_REGULATOR - /* Post 2.6.34 we will be able to get a callback when - * the regulators are disabled which we can use but - * for now just assume that the power will be cut if - * the regulator API is in use. - */ - codec->cache_sync = 1; -#endif + regcache_cache_only(wm8904->regmap, true); + regcache_mark_dirty(wm8904->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); @@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); struct wm8904_pdata *pdata = wm8904->pdata; - u16 *reg_cache = codec->reg_cache; int ret, i; - codec->cache_sync = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) goto err_enable; } + regcache_cache_only(wm8904->regmap, true); /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_VU, WM8904_ADC_VU); @@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec) if (!pdata->gpio_cfg[i]) continue; - reg_cache[WM8904_GPIO_CONTROL_1 + i] - = pdata->gpio_cfg[i] & 0xffff; + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + pdata->gpio_cfg[i]); } /* Zero is the default value for these anyway */ for (i = 0; i < WM8904_MIC_REGS; i++) - reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] - = pdata->mic_cfg[i]; + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + pdata->mic_cfg[i]); } /* Set Class W by default - this will be managed by the Class diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index aa8c98b628da..1436b6ce74d1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; - if (!wm8994->jackdet || !wm8994->jack_cb) - return; - if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8af422e38fd0..dc9b42b7fc4d 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - regcache_cache_only(codec->control_data, true); - /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, @@ -3051,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; } @@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - ret = wm8996_reset(wm8996); if (ret < 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); goto err_regmap; } + regcache_cache_only(wm8996->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 1c2aa7fab3fd..4da5fc55c7ee 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -33,7 +33,6 @@ #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + if (ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ } else { @@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp) u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + if (ssp->type == PXA25x_SSP) div = ((sscr0 >> 8) & 0xff) * 2 + 2; else div = ((sscr0 >> 8) & 0xfff) + 1; @@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_PLL: /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) + if (ssp->type == PXA25x_SSP) priv->sysclk = 1843200; else priv->sysclk = 13000000; @@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_disable(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_enable(ssp->clk); return 0; @@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, case PXA_SSP_AUDIO_DIV_SCDB: val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val &= ~SSACD_SCDX8; -#endif switch (div) { case PXA_SSP_CLK_SCDB_1: val |= SSACD_SCDB; break; case PXA_SSP_CLK_SCDB_4: break; -#if defined(CONFIG_PXA3xx) case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val |= SSACD_SCDX8; else return -EINVAL; break; -#endif default: return -EINVAL; } @@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif switch (freq_out) { case 5622000: @@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, break; default: -#ifdef CONFIG_PXA3xx /* PXA3xx has a clock ditherer which can be used to generate * a wider range of frequencies - calculate a value for it. */ - if (cpu_is_pxa3xx()) { + if (ssp->type == PXA3xx_SSP) { u32 val; u64 tmp = 19968; tmp *= 1000000; @@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val, freq_out); break; } -#endif return -EINVAL; } @@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) sscr0 |= SSCR0_FPCKE; -#endif sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: @@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * trying and failing a lot; some of the registers * needed for that mode are only available on PXA3xx. */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) return -EINVAL; sspsp |= SSPSP_SFRMWDTH(width * 2); @@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_EDMYSTOP(3); sspsp |= SSPSP_DMYSTOP(3); sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif } else { /* The frame width is the width the LRCLK is * asserted for; the delay is expressed in diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0b0df49d9d33..3b6da91188a9 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_wm8903_remove(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + wm8903_mic_detect(codec, NULL, 0, 0); + + return 0; +} + static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .dai_link = &tegra_wm8903_dai, .num_links = 1, + .remove = tegra_wm8903_remove, + .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715ab32fe..56ad923bf6b5 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6906901debb..0f647d22cb4a 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -414,7 +414,7 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, { struct list_head *p; struct snd_usb_endpoint *ep; - int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&chip->mutex); @@ -434,16 +434,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", ep_num); - /* select the alt setting once so the endpoints become valid */ - ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber, - alts->desc.bAlternateSetting); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - ep = NULL; - goto __exit_unlock; - } - ep = kzalloc(sizeof(*ep), GFP_KERNEL); if (!ep) goto __exit_unlock; @@ -831,9 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) - return -EINVAL; - /* just to be sure */ deactivate_urbs(ep, 0, 1); wait_clear_urbs(ep); @@ -911,9 +898,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (snd_BUG_ON(ep->use_count == 0)) return; - if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) - return; - if (--ep->use_count == 0) { deactivate_urbs(ep, force, can_sleep); ep->data_subs = NULL; @@ -927,42 +911,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, } /** - * snd_usb_endpoint_activate: activate an snd_usb_endpoint - * - * @ep: the endpoint to activate - * - * If the endpoint is not currently in use, this functions will select the - * correct alternate interface setting for the interface of this endpoint. - * - * In case of any active users, this functions does nothing. - * - * Returns an error if usb_set_interface() failed, 0 in all other - * cases. - */ -int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) -{ - if (ep->use_count != 0) - return 0; - - if (!ep->chip->shutdown && - !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) { - int ret; - - ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx); - if (ret < 0) { - snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", - __func__, ret); - clear_bit(EP_FLAG_ACTIVATED, &ep->flags); - return ret; - } - - return 0; - } - - return -EBUSY; -} - -/** * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint * * @ep: the endpoint to deactivate @@ -980,24 +928,15 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) if (!ep) return -EINVAL; + deactivate_urbs(ep, 1, 1); + wait_clear_urbs(ep); + if (ep->use_count != 0) return 0; - if (!ep->chip->shutdown && - test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) { - int ret; - - ret = usb_set_interface(ep->chip->dev, ep->iface, 0); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - return ret; - } + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); - return 0; - } - - return -EBUSY; + return 0; } /** diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 41daaa24c25f..e71fe55cebef 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -341,6 +341,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = audigy2nx_map, .selector_map = audigy2nx_selectors, }, + { /* Logitech, Inc. QuickCam Pro for Notebooks */ + .id = USB_ID(0x046d, 0x0991), + .ignore_ctl_error = 1, + }, + { /* Logitech, Inc. QuickCam E 3500 */ + .id = USB_ID(0x046d, 0x09a4), + .ignore_ctl_error = 1, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b7601973..a1298f379428 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -261,19 +261,6 @@ static void stop_endpoints(struct snd_usb_substream *subs, force, can_sleep, wait); } -static int activate_endpoints(struct snd_usb_substream *subs) -{ - if (subs->sync_endpoint) { - int ret; - - ret = snd_usb_endpoint_activate(subs->sync_endpoint); - if (ret < 0) - return ret; - } - - return snd_usb_endpoint_activate(subs->data_endpoint); -} - static int deactivate_endpoints(struct snd_usb_substream *subs) { int reta, retb; @@ -314,6 +301,33 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (fmt == subs->cur_audiofmt) return 0; + /* close the old interface */ + if (subs->interface >= 0 && subs->interface != fmt->iface) { + err = usb_set_interface(subs->dev, subs->interface, 0); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + subs->interface = -1; + subs->altset_idx = 0; + } + + /* set interface */ + if (subs->interface != fmt->iface || + subs->altset_idx != fmt->altset_idx) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + snd_printdd(KERN_INFO "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, fmt->endpoint, subs->direction, SND_USB_ENDPOINT_TYPE_DATA); @@ -354,17 +368,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || - ( is_playback && !implicit_fb))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } @@ -383,7 +401,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; } - if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) + if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) return err; subs->cur_audiofmt = fmt; @@ -446,7 +464,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct usb_interface *iface; iface = usb_ifnum_to_if(subs->dev, fmt->iface); alts = &iface->altsetting[fmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, subs->interface, alts, fmt, rate); + ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate); if (ret < 0) return ret; subs->cur_rate = rate; @@ -456,12 +474,6 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ stop_endpoints(subs, 0, 0, 0); - deactivate_endpoints(subs); - - ret = activate_endpoints(subs); - if (ret < 0) - goto unlock; - ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, subs->sync_endpoint); if (ret < 0) @@ -496,6 +508,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); stop_endpoints(subs, 0, 1, 1); + deactivate_endpoints(subs); mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -934,16 +947,20 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { - int ret; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, 0, 0, 0); - ret = deactivate_endpoints(subs); + + if (!as->chip->shutdown && subs->interface >= 0) { + usb_set_interface(subs->dev, subs->interface, 0); + subs->interface = -1; + } + subs->pcm_substream = NULL; snd_usb_autosuspend(subs->stream->chip); - return ret; + return 0; } /* Since a URB can handle only a single linear buffer, we must use double @@ -1147,7 +1164,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return -EINVAL; } -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) { int err; struct snd_usb_substream *subs = substream->runtime->private_data; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d89ab4c7d44b..79780fa57a43 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1831,6 +1831,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x014d), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "GT-100", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { |