Merge 3.10-rc5 into usb-next

We need the changes in this branch.

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

15 files changed, 99 insertions, 50 deletions

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ae85bbd2e6f8..4b1524a861f3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -788,6 +788,8 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable)
                 return;
         if (codec->inv_eapd)
                 enable = !enable;
+        if (spec->keep_eapd_on && !enable)
+                return;
         snd_hda_codec_update_cache(codec, pin, 0,
                                    AC_VERB_SET_EAPD_BTLENABLE,
                                    enable ? 0x02 : 0x00);
@@ -1938,17 +1940,7 @@ static int create_speaker_out_ctls(struct hda_codec *codec)
  * independent HP controls
  */
 
-/* update HP auto-mute state too */
-static void update_hp_automute_hook(struct hda_codec *codec)
-{
-        struct hda_gen_spec *spec = codec->spec;
-
-        if (spec->hp_automute_hook)
-                spec->hp_automute_hook(codec, NULL);
-        else
-                snd_hda_gen_hp_automute(codec, NULL);
-}
-
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack);
 static int indep_hp_info(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_info *uinfo)
 {
@@ -2009,7 +2001,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol,
                 else
                         *dacp = spec->alt_dac_nid;
 
-                update_hp_automute_hook(codec);
+                call_hp_automute(codec, NULL);
                 ret = 1;
         }
  unlock:
@@ -2305,7 +2297,7 @@ static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force)
                 else
                         val = PIN_HP;
                 set_pin_target(codec, pin, val, true);
-                update_hp_automute_hook(codec);
+                call_hp_automute(codec, NULL);
         }
 }
 
@@ -2714,7 +2706,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
                         val = snd_hda_get_default_vref(codec, nid);
         }
         snd_hda_set_pin_ctl_cache(codec, nid, val);
-        update_hp_automute_hook(codec);
+        call_hp_automute(codec, NULL);
 
         return 1;
 }
@@ -3859,20 +3851,42 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch);
 
-/* update jack retasking */
-static void update_automute_all(struct hda_codec *codec)
+/* call appropriate hooks */
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
 {
         struct hda_gen_spec *spec = codec->spec;
+        if (spec->hp_automute_hook)
+                spec->hp_automute_hook(codec, jack);
+        else
+                snd_hda_gen_hp_automute(codec, jack);
+}
 
-        update_hp_automute_hook(codec);
+static void call_line_automute(struct hda_codec *codec,
+                               struct hda_jack_tbl *jack)
+{
+        struct hda_gen_spec *spec = codec->spec;
         if (spec->line_automute_hook)
-                spec->line_automute_hook(codec, NULL);
+                spec->line_automute_hook(codec, jack);
         else
-                snd_hda_gen_line_automute(codec, NULL);
+                snd_hda_gen_line_automute(codec, jack);
+}
+
+static void call_mic_autoswitch(struct hda_codec *codec,
+                                struct hda_jack_tbl *jack)
+{
+        struct hda_gen_spec *spec = codec->spec;
         if (spec->mic_autoswitch_hook)
-                spec->mic_autoswitch_hook(codec, NULL);
+                spec->mic_autoswitch_hook(codec, jack);
         else
-                snd_hda_gen_mic_autoswitch(codec, NULL);
+                snd_hda_gen_mic_autoswitch(codec, jack);
+}
+
+/* update jack retasking */
+static void update_automute_all(struct hda_codec *codec)
+{
+        call_hp_automute(codec, NULL);
+        call_line_automute(codec, NULL);
+        call_mic_autoswitch(codec, NULL);
 }
 
 /*
@@ -4009,9 +4023,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
                 snd_printdd("hda-codec: Enable HP auto-muting on NID 0x%x\n",
                             nid);
                 snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_HP_EVENT,
-                                                    spec->hp_automute_hook ?
-                                                    spec->hp_automute_hook :
-                                                    snd_hda_gen_hp_automute);
+                                                    call_hp_automute);
                 spec->detect_hp = 1;
         }
 
@@ -4024,9 +4036,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
                                 snd_printdd("hda-codec: Enable Line-Out auto-muting on NID 0x%x\n", nid);
                                 snd_hda_jack_detect_enable_callback(codec, nid,
                                                                     HDA_GEN_FRONT_EVENT,
-                                                                    spec->line_automute_hook ?
-                                                                    spec->line_automute_hook :
-                                                                    snd_hda_gen_line_automute);
+                                                                    call_line_automute);
                                 spec->detect_lo = 1;
                         }
                 spec->automute_lo_possible = spec->detect_hp;
@@ -4068,9 +4078,7 @@ static bool auto_mic_check_imux(struct hda_codec *codec)
                 snd_hda_jack_detect_enable_callback(codec,
                                                     spec->am_entry[i].pin,
                                                     HDA_GEN_MIC_EVENT,
-                                                    spec->mic_autoswitch_hook ?
-                                                    spec->mic_autoswitch_hook :
-                                                    snd_hda_gen_mic_autoswitch);
+                                                    call_mic_autoswitch);
         return true;
 }
 
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 54e665160379..76200314ee95 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -222,6 +222,7 @@ struct hda_gen_spec {
         unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
         unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
         unsigned int own_eapd_ctl:1; /* set EAPD by own function */
+        unsigned int keep_eapd_on:1; /* don't turn off EAPD automatically */
         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 */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 59d2e91a9ab6..02e22b4458d2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
         SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
         SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -3530,6 +3532,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
+        SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
         SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e0dadcf2030d..e5245544eb52 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -136,6 +136,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
                 spec->codec_type = VT1708S;
         spec->no_pin_power_ctl = 1;
         spec->gen.indep_hp = 1;
+        spec->gen.keep_eapd_on = 1;
         spec->gen.pcm_playback_hook = via_playback_pcm_hook;
         return spec;
 }
@@ -231,9 +232,14 @@ static void vt1708_update_hp_work(struct hda_codec *codec)
 
 static void set_widgets_power_state(struct hda_codec *codec)
 {
+#if 0 /* FIXME: the assumed connections don't match always with the
+       * actual routes by the generic parser, so better to disable
+       * the control for safety.
+       */
         struct via_spec *spec = codec->spec;
         if (spec->set_widgets_power_state)
                 spec->set_widgets_power_state(codec);
+#endif
 }
 
 static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
@@ -478,7 +484,9 @@ static int via_suspend(struct hda_codec *codec)
                 /* Fix pop noise on headphones */
                 int i;
                 for (i = 0; i < spec->gen.autocfg.hp_outs; i++)
-                        snd_hda_set_pin_ctl(codec, spec->gen.autocfg.hp_pins[i], 0);
+                        snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i],
+                                            0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                            0x00);
         }
 
         return 0;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index d59abe1682c5..748e82d4d257 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card,
         if (rc)
                 goto error_out;
 
-        if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) {
+        rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30));
+        if (rc < 0) {
                 dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA");
                 goto error_out_enabled;
         }
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec09..030f53c96ec0 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
         { CS42L52_BEEP_VOL, 0x00 },     /* r1D Beep Volume off Time */
         { CS42L52_BEEP_TONE_CTL, 0x00 },        /* r1E Beep Tone Cfg. */
         { CS42L52_TONE_CTL, 0x00 },     /* r1F Tone Ctl */
-        { CS42L52_MASTERA_VOL, 0x88 },  /* r20 Master A Volume */
+        { CS42L52_MASTERA_VOL, 0x00 },  /* r20 Master A Volume */
         { CS42L52_MASTERB_VOL, 0x00 },  /* r21 Master B Volume */
         { CS42L52_HPA_VOL, 0x00 },      /* r22 Headphone A Volume */
         { CS42L52_HPB_VOL, 0x00 },      /* r23 Headphone B Volume */
@@ -225,7 +225,7 @@ static const char * const mic_bias_level_text[] = {
 };
 
 static const struct soc_enum mic_bias_level_enum =
-        SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+        SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
                         ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
 static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -413,7 +413,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
         SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
         SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-                              CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+                              CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
 
         SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
                               CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +441,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
         SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
                             CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-                                6, 0x7f, 0x19, hl_tlv),
+                                0, 0x7f, 0x19, hl_tlv),
         SOC_DOUBLE_R("PCM Mixer Switch",
                      CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c059071..4277012c4719 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
 #define CS42L52_PB_CTL1_INV_PCMA                (1 << 2)
 #define CS42L52_PB_CTL1_MSTB_MUTE               (1 << 1)
 #define CS42L52_PB_CTL1_MSTA_MUTE               (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK               0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK               0x03
 #define CS42L52_PB_CTL1_MUTE                    3
 #define CS42L52_PB_CTL1_UNMUTE                  0
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c97..8d14a76c7249 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
         dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
         ret = request_threaded_irq(max98090->irq, NULL,
-                max98090_interrupt, IRQF_TRIGGER_FALLING,
+                max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
                 "max98090_interrupt", codec);
         if (ret < 0) {
                 dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e04776..ba38f0679662 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -976,6 +976,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
         if (ret != 0)
                 return ret;
 
+        arizona_init_spk(codec);
+
         snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
         priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152cb1097..dfd997aaadfc 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
         int drc = wm8994_get_drc(kcontrol->id.name);
 
+        if (drc < 0)
+                return drc;
         ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
 
         return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
         int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
+        if (block < 0)
+                return block;
+
         ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
 
         return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 {
         struct snd_soc_codec *codec = w->codec;
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-        struct wm8994 *control = codec->control_data;
+        struct wm8994 *control = wm8994->wm8994;
         int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
         int i;
         int dac;
@@ -3833,6 +3838,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
                         dev_dbg(codec->dev, "Ignoring removed jack\n");
                         return IRQ_HANDLED;
                 }
+        } else if (!(reg & WM8958_MICD_STS)) {
+                snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+                                    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+                                    wm8994->btn_mask);
+                goto out;
         }
 
         if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e9..81490febac6d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                                        int word_length)
 {
         u32 fmt;
-        u32 rotate = (word_length / 4) & 0x7;
+        u32 tx_rotate = (word_length / 4) & 0x7;
+        u32 rx_rotate = (32 - word_length) / 4;
         u32 mask = (1ULL << word_length) - 1;
 
         /*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
                                 TXSSZ(fmt), TXSSZ(0x0F));
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
-                                TXROT(rotate), TXROT(7));
+                                TXROT(tx_rotate), TXROT(7));
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
-                                RXROT(rotate), RXROT(7));
+                                RXROT(rx_rotate), RXROT(7));
                 mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
                                 mask);
         }
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f28..06a8000aa07b 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
                         goto err;
         }
 
-        snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-                                SND_SOC_DAPM_STREAM_START);
+        if (cstream->direction == SND_COMPRESS_PLAYBACK)
+                snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+                                        SND_SOC_DAPM_STREAM_START);
+        else
+                snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+                                        SND_SOC_DAPM_STREAM_START);
 
         /* cancel any delayed stream shutdown that is pending */
         rtd->pop_wait = 0;
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a1d9b0792a1e..b9defcdeb7ef 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
         0x94, 0x01, 0x5c, 0x02  /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
 };
 
-static const u8 known_fw_versions[][4] = {
-        { 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+        { 0x03, 0x01 }
 };
 
 struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
         int i;
 
         for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
-                if (!memcmp(version, known_fw_versions + i, 4))
+                if (!memcmp(version, known_fw_versions + i, 2))
                         return 0;
 
         snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ca4739c3f650..e5c7f9f20fdd 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -886,6 +886,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
         case USB_ID(0x046d, 0x0808):
         case USB_ID(0x046d, 0x0809):
         case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
+        case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
         case USB_ID(0x046d, 0x0991):
         /* Most audio usb devices lie about volume resolution.
          * Most Logitech webcams have res = 384.
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 7f1722f82c89..8b75bcf136f6 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -215,7 +215,13 @@
         .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 },
 {
-        USB_DEVICE(0x046d, 0x0990),
+        .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+                       USB_DEVICE_ID_MATCH_INT_CLASS |
+                       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+        .idVendor = 0x046d,
+        .idProduct = 0x0990,
+        .bInterfaceClass = USB_CLASS_AUDIO,
+        .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
         .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
                 .vendor_name = "Logitech, Inc.",
                 .product_name = "QuickCam Pro 9000",
@@ -1792,7 +1798,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
         .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
                 .ifnum = 0,
-                .type = QUIRK_MIDI_STANDARD_INTERFACE
+                .type = QUIRK_MIDI_FIXED_ENDPOINT,
+                .data = & (const struct snd_usb_midi_endpoint_info) {
+                        .out_cables = 0x0007,
+                        .in_cables  = 0x0007
+                }
         }
 },
 {