Merge tag 'sound-4.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "This is a round of HD-audio fixes: there are a long-standing
  regression fix and a few more device/codec-specific quirks.

  In addition, a couple of FireWire regression fixes, a USB-audio quirk
  for Roland UA-22 and a sanity check in API for user-defined control
  elements"

* tag 'sound-4.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Don't access stereo amps for mono channel widgets
  ALSA: hda - Add workaround for MacBook Air 5,2 built-in mic
  ALSA: hda - Set single_adc_amp flag for CS420x codecs
  ALSA: snd-usb: add quirks for Roland UA-22
  ALSA: control: Add sanity checks for user ctl id name string
  ALSA: hda - Fix built-in mic on Compaq Presario CQ60
  ALSA: firewire-lib: leave unit reference counting completely
  Revert "ALSA: dice: fix wrong offsets for Dice interface"
  ALSA: hda - Fix regression of HD-audio controller fallback modes

diff --git a/sound/core/control.c b/sound/core/control.c
index 35324a8e83c867f126256f67b493b5320b9652c7..eeb691d1911f5716bb09b8eecacd24215a31564f 100644 (file)
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 
        if (info->count < 1)
                return -EINVAL;
+       if (!*info->id.name)
+               return -EINVAL;
+       if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name))
+               return -EINVAL;
        access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
                (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
                                 SNDRV_CTL_ELEM_ACCESS_INACTIVE|

diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index de7602bd69b569176ab5b41ab9065a0e04c7688a..27b044f84c816f5b4f15efa9782b7c4fbd1a217d 100644 (file)
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -298,24 +298,24 @@
  */
 #define RX_ISOCHRONOUS                 0x008
 
+/*
+ * Index of first quadlet to be interpreted; read/write.  If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START                   0x00c
+
 /*
  * The number of audio channels; read-only.  There will be one quadlet per
  * channel.
  */
-#define RX_NUMBER_AUDIO                        0x00c
+#define RX_NUMBER_AUDIO                        0x010
 
 /*
  * The number of MIDI ports, 0-8; read-only.  If > 0, there will be one
  * additional quadlet in each data block, following the audio quadlets.
  */
-#define RX_NUMBER_MIDI                 0x010
-
-/*
- * Index of first quadlet to be interpreted; read/write.  If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START                   0x014
+#define RX_NUMBER_MIDI                 0x014
 
 /*
  * Names of all audio channels; read-only.  Quadlets are byte-swapped.  Names

diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index ecfe20fd4de57868cefff26e8317ec54f180509a..f5c1d1bced59fee2e0012d9a5f96b85553bb53f8 100644 (file)
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
                } tx;
                struct {
                        u32 iso;
+                       u32 seq_start;
                        u32 number_audio;
                        u32 number_midi;
-                       u32 seq_start;
                        char names[RX_NAMES_SIZE];
                        u32 ac3_caps;
                        u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
                        break;
                snd_iprintf(buffer, "rx %u:\n", stream);
                snd_iprintf(buffer, "  iso channel: %d\n", (int)buf.rx.iso);
+               snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                snd_iprintf(buffer, "  audio channels: %u\n",
                            buf.rx.number_audio);
                snd_iprintf(buffer, "  midi ports: %u\n", buf.rx.number_midi);
-               snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                if (quadlets >= 68) {
                        dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
                        snd_iprintf(buffer, "  names: %s\n", buf.rx.names);

diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
index 5f17b77ee15222ad0e43c3c6b5b13688b386417f..f0e4d502d60482ae8374cf9f830b739eb9e38753 100644 (file)
--- a/sound/firewire/iso-resources.c
+++ b/sound/firewire/iso-resources.c
@@ -26,7 +26,7 @@
 int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 {
        r->channels_mask = ~0uLL;
-       r->unit = fw_unit_get(unit);
+       r->unit = unit;
        mutex_init(&r->mutex);
        r->allocated = false;
 
@@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r)
 {
        WARN_ON(r->allocated);
        mutex_destroy(&r->mutex);
-       fw_unit_put(r->unit);
 }
 EXPORT_SYMBOL(fw_iso_resources_destroy);
 

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a2ce773bdc624172b399afbb9207ed970b4c139f..17c2637d842c1c366275683f30d2a3cb743560b6 100644 (file)
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
                }
        }
 
-       if (!bus->no_response_fallback)
+       if (bus->no_response_fallback)
                return -1;
 
        if (!chip->polling_mode && chip->poll_count < 2) {

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4ec63313c8b1152390dbbf602018a212952..fe18071bf93aab4d44e779f9b7daaca3a2815b5b 100644 (file)
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
 {
        unsigned int caps = query_amp_caps(codec, nid, dir);
        int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
-       snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+       if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+               snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+       else
+               snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+                     unsigned int mask, unsigned int val)
+{
+       if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+               return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+                                               mask, val);
+       else
+               return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+                                               mask, val);
 }
 
 /* calculate amp value mask we can modify;
@@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
                return;
 
        val &= mask;
-       snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+       update_amp(codec, nid, dir, idx, mask, val);
 }
 
 static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
        has_amp = nid_has_mute(codec, mix, HDA_INPUT);
        for (i = 0; i < nums; i++) {
                if (has_amp)
-                       snd_hda_codec_amp_stereo(codec, mix,
-                                                HDA_INPUT, i,
-                                                0xff, HDA_AMP_MUTE);
+                       update_amp(codec, mix, HDA_INPUT, i,
+                                  0xff, HDA_AMP_MUTE);
                else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-                       snd_hda_codec_amp_stereo(codec, conn[i],
-                                                HDA_OUTPUT, 0,
-                                                0xff, HDA_AMP_MUTE);
+                       update_amp(codec, conn[i], HDA_OUTPUT, 0,
+                                  0xff, HDA_AMP_MUTE);
        }
 }
 

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 1589c9bcce3e15a230f87d352f2ae165252c358d..dd2b3d92071f698f41a75d2d7b7877eb6b357c00 100644 (file)
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
        SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
        SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
        SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+       SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
        SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
        SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
        {} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
                return -ENOMEM;
 
        spec->gen.automute_hook = cs_automute;
+       codec->single_adc_amp = 1;
 
        snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
                           cs420x_fixups);

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fd3ed18670e9c4005d115a26f6410c7efd4faff8..da67ea8645a6e8462f23b8f8b1e96044e66812f7 100644 (file)
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -223,6 +223,7 @@ enum {
        CXT_PINCFG_LENOVO_TP410,
        CXT_PINCFG_LEMOTE_A1004,
        CXT_PINCFG_LEMOTE_A1205,
+       CXT_PINCFG_COMPAQ_CQ60,
        CXT_FIXUP_STEREO_DMIC,
        CXT_FIXUP_INC_MIC_BOOST,
        CXT_FIXUP_HEADPHONE_MIC_PIN,
@@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = {
                .type = HDA_FIXUP_PINS,
                .v.pins = cxt_pincfg_lemote,
        },
+       [CXT_PINCFG_COMPAQ_CQ60] = {
+               .type = HDA_FIXUP_PINS,
+               .v.pins = (const struct hda_pintbl[]) {
+                       /* 0x17 was falsely set up as a mic, it should 0x1d */
+                       { 0x17, 0x400001f0 },
+                       { 0x1d, 0x97a70120 },
+                       { }
+               }
+       },
        [CXT_FIXUP_STEREO_DMIC] = {
                .type = HDA_FIXUP_FUNC,
                .v.func = cxt_fixup_stereo_dmic,
@@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = {
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
+       SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60),
        SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
        {}
 };

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 67d476548dcf9094acd61e0ff74f0f320af1556a..07f984d5f5162809ee6124f651585df5c3a4496f 100644 (file)
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
                }
        }
 },
+{
+       USB_DEVICE(0x0582, 0x0159),
+       .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+               /* .vendor_name = "Roland", */
+               /* .product_name = "UA-22", */
+               .ifnum = QUIRK_ANY_INTERFACE,
+               .type = QUIRK_COMPOSITE,
+               .data = (const struct snd_usb_audio_quirk[]) {
+                       {
+                               .ifnum = 0,
+                               .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                       },
+                       {
+                               .ifnum = 1,
+                               .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                       },
+                       {
+                               .ifnum = 2,
+                               .type = QUIRK_MIDI_FIXED_ENDPOINT,
+                               .data = & (const struct snd_usb_midi_endpoint_info) {
+                                       .out_cables = 0x0001,
+                                       .in_cables = 0x0001
+                               }
+                       },
+                       {
+                               .ifnum = -1
+                       }
+               }
+       }
+},
 /* this catches most recent vendor-specific Roland devices */
 {
        .match_flags = USB_DEVICE_ID_MATCH_VENDOR |