diff options
29 files changed, 306 insertions, 140 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3fd1a7e2492..552b97afbca 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) sdev->pcmid = -1; list_del(&ldev->list); layouts_list_items--; + kfree(ldev); outnodev: of_node_put(sound); layout_device = NULL; - kfree(ldev); return -ENODEV; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86d0caf91b3..62e90b862a0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t avail = 0; long wait_time, tout; + init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); + add_wait_queue(&runtime->tsleep, &wait); + if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { @@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } wait_time = msecs_to_jiffies(wait_time * 1000); } - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->tsleep, &wait); + for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; break; } + + /* + * We need to check if space became available already + * (and thus the wakeup happened already) first to close + * the race of space already having become available. + * This check must happen after been added to the waitqueue + * and having current state be INTERRUPTIBLE. + */ + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->twake) + break; snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout_interruptible(wait_time); + + tout = schedule_timeout(wait_time); + snd_pcm_stream_lock_irq(substream); + set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; @@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, err = -EIO; break; } - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->twake) - break; } _endloop: + set_current_state(TASK_RUNNING); remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; diff --git a/sound/core/timer.c b/sound/core/timer.c index 7c1cbf0a0dc..67ebf1c21c0 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -328,6 +328,8 @@ int snd_timer_close(struct snd_timer_instance *timeri) mutex_unlock(®ister_mutex); } else { timer = timeri->timer; + if (snd_BUG_ON(!timer)) + goto out; /* wait, until the active callback is finished */ spin_lock_irq(&timer->lock); while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { @@ -353,6 +355,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) } mutex_unlock(®ister_mutex); } + out: if (timeri->private_free) timeri->private_free(timeri); kfree(timeri->owner); @@ -531,6 +534,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri) if (err < 0) return err; timer = timeri->timer; + if (!timer) + return -EINVAL; spin_lock_irqsave(&timer->lock, flags); timeri->cticks = timeri->ticks; timeri->pticks = 0; diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c index 8f7d175767a..6f13ab4afc6 100644 --- a/sound/oss/pas2_pcm.c +++ b/sound/oss/pas2_pcm.c @@ -63,13 +63,13 @@ static int pcm_set_speed(int arg) if (pcm_channels & 2) { - foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg; - arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo; + foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg; + arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo; } else { - foo = (CLOCK_TICK_RATE + (arg / 2)) / arg; - arg = (CLOCK_TICK_RATE + (foo / 2)) / foo; + foo = (PIT_TICK_RATE + (arg / 2)) / arg; + arg = (PIT_TICK_RATE + (foo / 2)) / foo; } pcm_speed = arg; diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 9b800ce5100..2fc0624024b 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -673,7 +673,8 @@ static void configure_nonsound_components(void) if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */ printk(KERN_INFO "PSS: CDROM port not enabled.\n"); - } else if (check_region(pss_cdrom_port, 2)) { + } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) { + pss_cdrom_port = -1; printk(KERN_ERR "PSS: CDROM I/O port conflict.\n"); } else { set_io_base(devc, CONF_CDROM, pss_cdrom_port); @@ -1232,7 +1233,8 @@ static void __exit cleanup_pss(void) if(pssmpu) unload_pss_mpu(&cfg_mpu); unload_pss(&cfg); - } + } else if (pss_cdrom_port != -1) + release_region(pss_cdrom_port, 2); if(!pss_keep_settings) /* Keep hardware settings if asked */ { diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 50abf5bf8e0..88168044375 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,5 +1,10 @@ # ALSA PCI drivers +config SND_TEA575X + tristate + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 + menuconfig SND_PCI bool "PCI sound devices" depends on PCI @@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and SF64-PCR) into the snd-fm801 driver. -config SND_TEA575X - tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 - source "sound/pci/hda/Kconfig" config SND_HDSP diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 200c9a1d48b..a872d0a8297 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = { 0x103c0944, /* HP nc6220 */ 0x103c0934, /* HP nc8220 */ 0x103c006d, /* HP nx9105 */ + 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */ 0x17340088, /* FSC Scenic-W */ 0 /* end */ }; diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 65b7ca13115..bd47521b24e 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count, if (!p_cache) return NULL; - p_cache->p_info = - kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL); + p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count, + GFP_KERNEL); if (!p_cache->p_info) { kfree(p_cache); return NULL; } - memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count); p_cache->cache_size_in_bytes = size_in_bytes; p_cache->control_count = control_count; p_cache->p_cache = p_dsp_control_buffer; diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e4d76a270c9..579fc0dce12 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) int err; snd_azf3328_dbgcallenter(); - if (dev >= SNDRV_CARDS) - return -ENODEV; + if (dev >= SNDRV_CARDS) { + err = -ENODEV; + goto out; + } if (!enable[dev]) { dev++; - return -ENOENT; + err = -ENOENT; + goto out; } err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) - return err; + goto out; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index f9123f09e83..32b02d90670 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); +#define TUNER_DISABLED (1<<3) #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip) __end_hw: #ifdef CONFIG_SND_FM801_TEA575X_BOOL - snd_tea575x_exit(&chip->tea); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) + snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1236,7 +1238,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { @@ -1251,11 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card, } if (tea575x_tuner == 4) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); - return -ENODEV; + chip->tea575x_tuner = TUNER_DISABLED; } } - strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) { + strlcpy(chip->tea.card, + snd_fm801_tea575x_gpios[(tea575x_tuner & + TUNER_TYPE_MASK) - 1].name, + sizeof(chip->tea.card)); + } #endif *rchip = chip; diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c index be58bf2f3ae..2e5876ce71f 100644 --- a/sound/pci/hda/alc268_quirks.c +++ b/sound/pci/hda/alc268_quirks.c @@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { static const struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, @@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_toshiba_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_DMIC] = { - .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_ASPIRE_ONE] = { - .mixers = { alc268_acer_aspire_one_mixer, - alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ZEPTO] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = { }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { - .mixers = { alc268_test_mixer, alc268_capture_mixer }, + .mixers = { alc268_test_mixer }, + .cap_mixer = alc268_capture_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_volume_init_verbs, alc268_beep_init_verbs }, diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c index 14fdcf29b15..5ac0e2162a4 100644 --- a/sound/pci/hda/alc269_quirks.c +++ b/sound/pci/hda/alc269_quirks.c @@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC), SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC), - SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", - ALC269_DMIC), - SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", - ALC269_DMIC), - SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e7850c238c..f3aefef3721 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, return -1; } recursive++; - for (i = 0; i < nums; i++) + for (i = 0; i < nums; i++) { + unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); + if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) + continue; if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) return i; + } return -1; } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d09c3..c34f730f481 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = { SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ }; -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, +static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, int byte_index) { unsigned int val; val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); - #ifdef BE_PARANOID printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); #endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; + return val; } #define GRAB_BITS(buf, byte, lowbit, bits) \ @@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, if (!buf) return -ENOMEM; - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, nid, i); + for (i = 0; i < size; i++) { + unsigned int val = hdmi_get_eld_data(codec, nid, i); + if (!(val & AC_ELDD_ELD_VALID)) { + if (!i) { + snd_printd(KERN_INFO + "HDMI: invalid ELD data\n"); + ret = -EINVAL; + goto error; + } + snd_printd(KERN_INFO + "HDMI: invalid ELD data byte %d\n", i); + val = 0; + } else + val &= AC_ELDD_ELD_DATA; + buf[i] = val; + } ret = hdmi_update_eld(eld, buf, size); +error: kfree(buf); return ret; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 47d6ffc9b5b..c45f3e69bcf 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, unsigned int *idxp) { - int i; + int i, idx; hda_nid_t nid; nid = codec->start_nid; @@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; - *idxp = snd_hda_get_conn_index(codec, nid, pin, false); - if (*idxp >= 0) + idx = snd_hda_get_conn_index(codec, nid, pin, false); + if (idx >= 0) { + *idxp = idx; return nid; + } } return 0; } @@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 502fc949945..7696d05b935 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin, #define MAX_AUTO_DACS 5 +#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */ + /* fill analog DAC list from the widget tree */ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) { @@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - nums++; + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { + filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { hda_nid_t dac = spec->dac_info[i].dac; - if (!dac) + if (!dac || (dac & DAC_SLAVE_FLAG)) continue; switch (spec->dac_info[i].type) { case AUTO_PIN_LINE_OUT: @@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) } if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items) cx_auto_check_auto_mic(codec); - if (imux->num_items > 1 && !spec->auto_mic) { + if (imux->num_items > 1) { for (i = 1; i < imux->num_items; i++) { if (spec->imux_info[i].adc != spec->imux_info[0].adc) { spec->adc_switching = 1; @@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec) nid = spec->dac_info[i].dac; if (!nid) nid = spec->multiout.dac_nids[0]; + else if (nid & DAC_SLAVE_FLAG) + nid &= ~DAC_SLAVE_FLAG; select_connection(codec, spec->dac_info[i].pin, nid); } if (spec->auto_mute) { @@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) for (i = 0; i < spec->dac_info_filled; i++) { const char *label; int idx, type; - if (!spec->dac_info[i].dac) - continue; + hda_nid_t dac = spec->dac_info[i].dac; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; @@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) idx = num_spk++; break; } - err = try_add_pb_volume(codec, spec->dac_info[i].dac, + err = try_add_pb_volume(codec, dac, spec->dac_info[i].pin, label, idx); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e125c60fe35..0503c999e7d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -168,7 +168,7 @@ struct alc_spec { unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ unsigned int automute:1; /* HP automute enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ - unsigned int automute_lines:1; /* automute line-out as well */ + unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */ unsigned int automute_hp_lo:1; /* both HP and LO available */ /* other flags */ @@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute_lines || !spec->automute) + if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines)) on = 0; else on = spec->jack_present; @@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute) - return; spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); + if (!spec->automute) + return; update_speakers(codec); } @@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute || !spec->detect_line) - return; spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); + if (!spec->automute || !spec->detect_line) + return; update_speakers(codec); } @@ -803,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, unsigned int val; if (!spec->automute) val = 0; - else if (!spec->automute_lines) + else if (!spec->automute_hp_lo || !spec->automute_lines) val = 1; else val = 2; @@ -824,7 +824,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, spec->automute = 0; break; case 1: - if (spec->automute && !spec->automute_lines) + if (spec->automute && + (!spec->automute_hp_lo || !spec->automute_lines)) return 0; spec->automute = 1; spec->automute_lines = 0; @@ -1784,6 +1785,7 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "PCM Playback Volume", NULL, }; @@ -1798,6 +1800,7 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "PCM Playback Switch", NULL, }; @@ -3081,16 +3084,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin, dac; pin = spec->autocfg.hp_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.hp_nid); + if (pin) { + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, - spec->multiout.extra_out_nid[0]); + if (pin) { + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } /* @@ -4484,6 +4493,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec, spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; } +static void alc269_fixup_stereo_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + int coef; + + if (action != ALC_FIXUP_ACT_INIT) + return; + /* The digital-mic unit sends PDM (differential signal) instead of + * the standard PCM, thus you can't record a valid mono stream as is. + * Below is a workaround specific to ALC269 to control the dmic + * signal source as mono. + */ + coef = alc_read_coef_idx(codec, 0x07); + alc_write_coef_idx(codec, 0x07, coef | 0x80); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -4494,6 +4519,7 @@ enum { ALC275_FIXUP_SONY_HWEQ, ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, + ALC269_FIXUP_STEREO_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -4556,10 +4582,19 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, }, + [ALC269_FIXUP_STEREO_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_stereo_dmic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aa376b59c00..1b7c11432aa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac_vrefout_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { @@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +#endif static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) @@ -6571,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 84d8798bf33..4ebfbd874c9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec) struct via_spec *spec = codec->spec; struct nid_path *path; bool check_dac; - hda_nid_t pin, dac; + hda_nid_t pin, dac = 0; int err; pin = spec->autocfg.speaker_pins[0]; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6edc67ced90..493e3946756 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period) break; case MADIface: freq_const = 131072000000000ULL; + break; + default: + snd_BUG(); + return 0; } return div_u64(freq_const, period); @@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) switch (hdspm->io_type) { case MADIface: - n = 131072000000000ULL; /* 125 MHz */ - break; + n = 131072000000000ULL; /* 125 MHz */ + break; case MADI: case AES32: - n = 110069313433624ULL; /* 105 MHz */ - break; + n = 110069313433624ULL; /* 105 MHz */ + break; case RayDAT: case AIO: - n = 104857600000000ULL; /* 100 MHz */ - break; + n = 104857600000000ULL; /* 100 MHz */ + break; + default: + snd_BUG(); + return; } n = div_u64(n, rate); diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index d0d493ca28a..2cf87f5afed 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) for (i = 0; i < N_URBS; i++) { usb_kill_urb(dev->data_urbs_in[i]); - usb_kill_urb(dev->data_urbs_out[i]); + + if (test_bit(i, &dev->outurb_active_mask)) + usb_kill_urb(dev->data_urbs_out[i]); } + + dev->outurb_active_mask = 0; } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) @@ -612,8 +616,9 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; - struct urb *out; - int frame, len, send_it = 0, outframe = 0; + struct urb *out = NULL; + int i, frame, len, send_it = 0, outframe = 0; + size_t offset = 0; if (urb->status || !info) return; @@ -623,7 +628,17 @@ static void read_completed(struct urb *urb) if (!dev->streaming) return; - out = dev->data_urbs_out[info->index]; + /* find an unused output urb that is unused */ + for (i = 0; i < N_URBS; i++) + if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) { + out = dev->data_urbs_out[i]; + break; + } + + if (!out) { + log("Unable to find an output urb to use\n"); + goto requeue; + } /* read the recently received packet and send back one which has * the same layout */ @@ -634,7 +649,8 @@ static void read_completed(struct urb *urb) len = urb->iso_frame_desc[outframe].actual_length; out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; - out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; + out->iso_frame_desc[outframe].offset = offset; + offset += len; if (len > 0) { spin_lock(&dev->spinlock); @@ -650,11 +666,15 @@ static void read_completed(struct urb *urb) } if (send_it) { - out->number_of_packets = FRAMES_PER_URB; + out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); + } else { + struct snd_usb_caiaq_cb_info *oinfo = out->context; + clear_bit(oinfo->index, &dev->outurb_active_mask); } +requeue: /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; @@ -676,6 +696,8 @@ static void write_completed(struct urb *urb) dev->output_running = 1; wake_up(&dev->prepare_wait_queue); } + + clear_bit(info->index, &dev->outurb_active_mask); } static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) @@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) if (!dev->data_cb_info) return -ENOMEM; + dev->outurb_active_mask = 0; + BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8)); + for (i = 0; i < N_URBS; i++) { dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index b2b310194ff..3f9c6339ae9 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -96,6 +96,7 @@ struct snd_usb_caiaqdev { int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; unsigned int samplerates, bpp; + unsigned long outurb_active_mask; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 4432ef7a70a..a213813487b 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, KEY_5, KEY_6, KEY_7 }; static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4, - KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 }; + KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 }; static unsigned short keycode_kore[] = { KEY_FN_F1, /* "menu" */ diff --git a/sound/usb/card.c b/sound/usb/card.c index 781d9e61adf..ed120ca2353 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -532,6 +532,7 @@ snd_usb_audio_probe(struct usb_device *dev, __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); + chip->probing = 0; mutex_unlock(®ister_mutex); __err_val: return NULL; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7c0d21ecd82..7d46e482375 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || - ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) { + ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c22fa76e363..cdd19d7fe50 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, if (p && p->dB) { cval->dBmin = p->dB->min; cval->dBmax = p->dB->max; + cval->initialized = 1; } } @@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, " Switch" : " Volume"); if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); - if (cval->dBmin < cval->dBmax) { + if (cval->dBmin < cval->dBmax || !cval->initialized) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | @@ -1191,6 +1192,11 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void if (state->mixer->protocol == UAC_VERSION_1) { csize = hdr->bControlSize; + if (!csize) { + snd_printdd(KERN_ERR "usbaudio: unit %u: " + "invalid bControlSize == 0\n", unitid); + return -EINVAL; + } channels = (hdr->bLength - 7) / csize - 1; bmaControls = hdr->bmaControls; } else { @@ -1934,15 +1940,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) struct mixer_build state; int err; const struct usbmix_ctl_map *map; - struct usb_host_interface *hostif; void *p; - hostif = mixer->chip->ctrl_intf; memset(&state, 0, sizeof(state)); state.chip = mixer->chip; state.mixer = mixer; - state.buffer = hostif->extra; - state.buflen = hostif->extralen; + state.buffer = mixer->hostif->extra; + state.buflen = mixer->hostif->extralen; /* check the mapping table */ for (map = usbmix_ctl_maps; map->id; map++) { @@ -1955,7 +1959,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } p = NULL; - while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { + while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen, + p, UAC_OUTPUT_TERMINAL)) != NULL) { if (mixer->protocol == UAC_VERSION_1) { struct uac1_output_terminal_descriptor *desc = p; @@ -2162,17 +2167,15 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { - struct usb_host_interface *hostif; struct usb_endpoint_descriptor *ep; void *transfer_buffer; int buffer_length; unsigned int epnum; - hostif = mixer->chip->ctrl_intf; /* we need one interrupt input endpoint */ - if (get_iface_desc(hostif)->bNumEndpoints < 1) + if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1) return 0; - ep = get_endpoint(hostif, 0); + ep = get_endpoint(mixer->hostif, 0); if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep)) return 0; @@ -2202,7 +2205,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, }; struct usb_mixer_interface *mixer; struct snd_info_entry *entry; - struct usb_host_interface *host_iface; int err; strcpy(chip->card->mixername, "USB Mixer"); @@ -2219,8 +2221,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, return -ENOMEM; } - host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; + switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) { case UAC_VERSION_1: default: mixer->protocol = UAC_VERSION_1; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ae1a14dcfe8..81b2d8a32fb 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,7 @@ struct usb_mixer_interface { struct snd_usb_audio *chip; + struct usb_host_interface *hostif; struct list_head list; unsigned int ignore_ctl_error; struct urb *urb; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index dba0b7f11c5..a42e3ef3832 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .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 */ { @@ -2417,6 +2451,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), .idProduct = 0x1020, }, +/* KeithMcMillen Stringport */ +{ + USB_DEVICE(0x1f38, 0x0001), + .bInterfaceClass = USB_CLASS_AUDIO, +}, + /* Miditech devices */ { USB_DEVICE(0x4752, 0x0011), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77762c99afb..81e07d84258 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) */ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { - int err, reg; + int err = 0, reg; int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { |