18 files changed, 91 insertions, 50 deletions

diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 108b643229ba..6205f37d547c 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
         struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
 
-        if (rtd && rtd->params)
+        if (rtd && rtd->params && rtd->params->drcmr)
                 *rtd->params->drcmr = 0;
 
         snd_pcm_set_runtime_buffer(substream, NULL);
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 1bcb360330e5..941f64a853eb 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -3,10 +3,6 @@
 # Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
 #
 
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-  obj-$(CONFIG_SND_SEQUENCER) += oss/
-endif
-
 snd-seq-device-objs := seq_device.o
 snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
                 seq_fifo.o seq_prioq.o seq_timer.o \
@@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o
 
 obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
 ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+  obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+  obj-$(CONFIG_SND_SEQUENCER) += oss/
 endif
 obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
 
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index edb11eefdfe3..2dcf45bf7293 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
                 if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
                         continue;
                 /* load real volume - better precision */
-                spin_lock_irqsave(&gus->reg_lock, flags);
+                spin_lock(&gus->reg_lock);
                 snd_gf1_select_voice(gus, pvoice->number);
                 snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
                 vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
                 snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
                 pcmp->final_volume = 1;
-                spin_unlock_irqrestore(&gus->reg_lock, flags);
+                spin_unlock(&gus->reg_lock);
         }
         spin_unlock_irqrestore(&gus->voice_alloc, flags);
         return change;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index f24bf1ecb36d..15e4138bce17 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
         .rate_max =             192000,
         .channels_min =         2,
         .channels_max =         2,
-        .buffer_bytes_max =     ((65536 - 64) * 8),
+        .buffer_bytes_max =     65536 - 128,
         .period_bytes_min =     64,
-        .period_bytes_max =     (65536 - 64),
+        .period_bytes_max =     32768 - 64,
         .periods_min =          2,
         .periods_max =          2,
         .fifo_size =            0,
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b3ead8..a7f4a671f7b7 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
 
         /* Allocate mem for amixer resource */
         amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
-        if (NULL == amixer) {
-                err = -ENOMEM;
-                return err;
-        }
+        if (!amixer)
+                return -ENOMEM;
 
         /* Check whether there are sufficient
          * amixer resources to meet request. */
+        err = 0;
         spin_lock_irqsave(&mgr->mgr_lock, flags);
         for (i = 0; i < desc->msr; i++) {
                 err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
 
         /* Allocate mem for sum resource */
         sum = kzalloc(sizeof(*sum), GFP_KERNEL);
-        if (NULL == sum) {
-                err = -ENOMEM;
-                return err;
-        }
+        if (!sum)
+                return -ENOMEM;
 
         /* Check whether there are sufficient sum resources to meet request. */
+        err = 0;
         spin_lock_irqsave(&mgr->mgr_lock, flags);
         for (i = 0; i < desc->msr; i++) {
                 err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c08c02..deb6cfa73600 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
 
 struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
         [LINEO1] = {.left = 0x40, .right = 0x41},
-        [LINEO2] = {.left = 0x70, .right = 0x71},
+        [LINEO2] = {.left = 0x60, .right = 0x61},
         [LINEO3] = {.left = 0x50, .right = 0x51},
-        [LINEO4] = {.left = 0x60, .right = 0x61},
+        [LINEO4] = {.left = 0x70, .right = 0x71},
         [LINEIM] = {.left = 0x45, .right = 0xc5},
         [SPDIFOO] = {.left = 0x00, .right = 0x01},
         [SPDIFIO] = {.left = 0x05, .right = 0x85},
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d8b702..df43a5cd3938 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
 
         /* Allocate mem for SRCIMP resource */
         srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
-        if (NULL == srcimp) {
-                err = -ENOMEM;
-                return err;
-        }
+        if (!srcimp)
+                return -ENOMEM;
 
         /* Check whether there are sufficient SRCIMP resources. */
+        err = 0;
         spin_lock_irqsave(&mgr->mgr_lock, flags);
         for (i = 0; i < desc->msr; i++) {
                 err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255de6beb..88480c0c58a0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
                                                   AC_VERB_GET_CONNECT_LIST, i);
                 range_val = !!(parm & (1 << (shift-1))); /* ranges */
                 val = parm & mask;
+                if (val == 0) {
+                        snd_printk(KERN_WARNING "hda_codec: "
+                                   "invalid CONNECT_LIST verb %x[%i]:%x\n",
+                                    nid, i, parm);
+                        return 0;
+                }
                 parm >>= shift;
                 if (range_val) {
                         /* ranges between the previous and this one */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index be7d25fa7f35..3da85caf8af1 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
         int mute = (!ucontrol->value.integer.value[0] &&
                     !ucontrol->value.integer.value[1]);
         /* toggle GPIO1 according to the mute state */
-        snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+        snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
                             mute ? 0x02 : 0x0);
         return ret;
 }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bbb9b42e2604..8c8b273116fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
                                               &dig_nid, 1);
                 if (err < 0)
                         continue;
+                if (dig_nid > 0x7f) {
+                        printk(KERN_ERR "alc880_auto: invalid dig_nid "
+                                "connection 0x%x for NID 0x%x\n", dig_nid,
+                                spec->autocfg.dig_out_pins[i]);
+                        continue;
+                }
                 if (!i)
                         spec->multiout.dig_out_nid = dig_nid;
                 else {
@@ -10625,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
         alc262_lenovo_3000_automute(codec, 1);
 }
 
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+                                  int dir, int idx, long *valp)
+{
+        int i, change = 0;
+
+        for (i = 0; i < 2; i++, valp++)
+                change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+                                                   HDA_AMP_MUTE,
+                                                   *valp ? 0 : HDA_AMP_MUTE);
+        return change;
+}
+
 /* bind hp and internal speaker mute (with plug check) */
 static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
                                          struct snd_ctl_elem_value *ucontrol)
@@ -10633,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
         long *valp = ucontrol->value.integer.value;
         int change;
 
-        change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-                                                 HDA_AMP_MUTE,
-                                                 valp ? 0 : HDA_AMP_MUTE);
-        change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-                                                 HDA_AMP_MUTE,
-                                                 valp ? 0 : HDA_AMP_MUTE);
-
+        change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+        change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
         if (change)
                 alc262_fujitsu_automute(codec, 0);
         return change;
@@ -10674,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
         long *valp = ucontrol->value.integer.value;
         int change;
 
-        change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-                                                 HDA_AMP_MUTE,
-                                                 valp ? 0 : HDA_AMP_MUTE);
-
+        change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
         if (change)
                 alc262_lenovo_3000_automute(codec, 0);
         return change;
@@ -11848,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
         long *valp = ucontrol->value.integer.value;
         int change;
 
-        change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                          HDA_AMP_MUTE,
-                                          valp[0] ? 0 : HDA_AMP_MUTE);
-        change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                           HDA_AMP_MUTE,
-                                           valp[1] ? 0 : HDA_AMP_MUTE);
+        change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
         if (change)
                 alc268_acer_automute(codec, 0);
         return change;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a18c1e..512f3b9b9a45 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2378,6 +2378,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
                       "Dell Vostro 1500", STAC_9205_DELL_M42),
         /* Gateway */
+        SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
         SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
         {} /* terminator */
 };
@@ -4065,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
         jack->nid = nid;
         jack->type = type;
 
-        sprintf(name, "%s at %s %s Jack",
+        snprintf(name, sizeof(name), "%s at %s %s Jack",
                 snd_hda_get_jack_type(def_conf),
                 snd_hda_get_jack_connectivity(def_conf),
                 snd_hda_get_jack_location(def_conf));
@@ -5854,6 +5855,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
 };
 
 static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+        SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+                           "Sony VAIO F/S", STAC_9872_VAIO),
         {} /* terminator */
 };
 
@@ -5866,6 +5869,8 @@ static int patch_stac9872(struct hda_codec *codec)
         if (spec == NULL)
                 return -ENOMEM;
         codec->spec = spec;
+        spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+        spec->pin_nids = stac9872_pin_nids;
 
         spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
                                                         stac9872_models,
@@ -5877,8 +5882,6 @@ static int patch_stac9872(struct hda_codec *codec)
                 stac92xx_set_config_regs(codec,
                                          stac9872_brd_tbl[spec->board_config]);
 
-        spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
-        spec->pin_nids = stac9872_pin_nids;
         spec->multiout.dac_nids = spec->dac_nids;
         spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
         spec->adc_nids = stac9872_adc_nids;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 235a71e5ac8d..b5ca02e2038c 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void)
         if (err < 0)
                 return err;
 #if defined(SUPPORT_JOYSTICK)
-        pci_register_driver(&joystick_driver);
+        err = pci_register_driver(&joystick_driver);
+        /* On failure unregister formerly registered audio driver */
+        if (err < 0)
+                pci_unregister_driver(&driver);
 #endif
-        return 0;
+        return err;
 }
 
 static void __exit alsa_card_riptide_exit(void)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ab099f482487..cb0d1bf34b57 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
         int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
         u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
         u16 pll_d = 1;
+        u8 reg;
 
         /* select data word length */
         data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
                 pll_q &= 0xf;
                 aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
                 aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
-        } else
+                /* disable PLL if it is bypassed */
+                reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+                aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+        } else {
                 aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+                /* enable PLL when it is used */
+                reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+                aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+        }
 
         /* Route Left DAC to left channel input and
          * right DAC to right channel input */
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 523aec188ccf..73525c048e7f 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
             * Native Instruments Kore Controller
             * Native Instruments Kore Controller 2
             * Native Instruments Audio Kontrol 1
+            * Native Instruments Audio 2 DJ
             * Native Instruments Audio 4 DJ
             * Native Instruments Audio 8 DJ
             * Native Instruments Guitar Rig Session I/O
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 8f9b60c5d74c..121af0644fd9 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
         case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
                 dev->samplerates |= SNDRV_PCM_RATE_192000;
                 /* fall thru */
+        case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
         case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
         case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
                 dev->samplerates |= SNDRV_PCM_RATE_88200;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index de38108f0b28..83e6c1312d47 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,13 +35,14 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
                          "{Native Instruments, RigKontrol3},"
                          "{Native Instruments, Kore Controller},"
                          "{Native Instruments, Kore Controller 2},"
                          "{Native Instruments, Audio Kontrol 1},"
+                         "{Native Instruments, Audio 2 DJ},"
                          "{Native Instruments, Audio 4 DJ},"
                          "{Native Instruments, Audio 8 DJ},"
                          "{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
                 .idVendor =     USB_VID_NATIVEINSTRUMENTS,
                 .idProduct =    USB_PID_AUDIO4DJ
         },
+        {
+                .match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+                .idVendor =     USB_VID_NATIVEINSTRUMENTS,
+                .idProduct =    USB_PID_AUDIO2DJ
+        },
         { /* terminator */ }
 };
 
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index ece73514854e..44e3edf88bef 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -10,6 +10,7 @@
 #define USB_PID_KORECONTROLLER  0x4711
 #define USB_PID_KORECONTROLLER2 0x4712
 #define USB_PID_AK1             0x0815
+#define USB_PID_AUDIO2DJ        0x041c
 #define USB_PID_AUDIO4DJ        0x0839
 #define USB_PID_AUDIO8DJ        0x1978
 #define USB_PID_SESSIONIO       0x1915
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c7b902358b7b..44b9cdc8a83b 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2661,7 +2661,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
         struct usb_interface_descriptor *altsd;
         int i, altno, err, stream;
         int format;
-        struct audioformat *fp;
+        struct audioformat *fp = NULL;
         unsigned char *fmt, *csep;
         int num;
 
@@ -2734,6 +2734,18 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                         continue;
                 }
 
+                /*
+                 * Blue Microphones workaround: The last altsetting is identical
+                 * with the previous one, except for a larger packet size, but
+                 * is actually a mislabeled two-channel setting; ignore it.
+                 */
+                if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
+                    fp && fp->altsetting == 1 && fp->channels == 1 &&
+                    fp->format == SNDRV_PCM_FORMAT_S16_LE &&
+                    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+                                                        fp->maxpacksize * 2)
+                        continue;
+
                 csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
                 /* Creamware Noah has this descriptor after the 2nd endpoint */
                 if (!csep && altsd->bNumEndpoints >= 2)