Merge branch 'fix/hda' into topic/hda

The fix for bitmap-overflow in Realtek codec driver is needed for the
further development of the auto-parser with badness evaluation.

7 files changed, 48 insertions, 12 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8e06eb96e11..0ffccc178957 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
         ALC_AUTOMUTE_MIXER,     /* mute/unmute mixer widget AMP */
 };
 
+#define MAX_VOL_NIDS    0x40
+
 struct alc_spec {
         /* codec parameterization */
         const struct snd_kcontrol_new *mixers[5];       /* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
         const hda_nid_t *capsrc_nids;
         hda_nid_t dig_in_nid;           /* digital-in NID; optional */
         hda_nid_t mixer_nid;            /* analog-mixer NID */
-        DECLARE_BITMAP(vol_ctls, 0x20 << 1);
-        DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+        DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+        DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
 
         /* capture setup for dynamic dual-adc switch */
         hda_nid_t cur_adc;
@@ -3125,7 +3127,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 static inline unsigned int get_ctl_pos(unsigned int data)
 {
         hda_nid_t nid = get_amp_nid_(data);
-        unsigned int dir = get_amp_direction_(data);
+        unsigned int dir;
+        if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+                return 0;
+        dir = get_amp_direction_(data);
         return (nid << 1) | dir;
 }
 
@@ -4399,6 +4404,7 @@ enum {
         ALC882_FIXUP_ACER_ASPIRE_8930G,
         ALC882_FIXUP_ASPIRE_8930G_VERBS,
         ALC885_FIXUP_MACPRO_GPIO,
+        ALC889_FIXUP_DAC_ROUTE,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -4452,6 +4458,23 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec,
         alc882_gpio_mute(codec, 1, 0);
 }
 
+/* Fix the connection of some pins for ALC889:
+ * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't
+ * work correctly (bko#42740)
+ */
+static void alc889_fixup_dac_route(struct hda_codec *codec,
+                                   const struct alc_fixup *fix, int action)
+{
+        if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+                hda_nid_t conn1[2] = { 0x0c, 0x0d };
+                hda_nid_t conn2[2] = { 0x0e, 0x0f };
+                snd_hda_override_conn_list(codec, 0x14, 2, conn1);
+                snd_hda_override_conn_list(codec, 0x15, 2, conn1);
+                snd_hda_override_conn_list(codec, 0x18, 2, conn2);
+                snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+        }
+}
+
 static const struct alc_fixup alc882_fixups[] = {
         [ALC882_FIXUP_ABIT_AW9D_MAX] = {
                 .type = ALC_FIXUP_PINS,
@@ -4599,6 +4622,10 @@ static const struct alc_fixup alc882_fixups[] = {
                 .type = ALC_FIXUP_FUNC,
                 .v.func = alc885_fixup_macpro_gpio,
         },
+        [ALC889_FIXUP_DAC_ROUTE] = {
+                .type = ALC_FIXUP_FUNC,
+                .v.func = alc889_fixup_dac_route,
+        },
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -4623,6 +4650,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
                       ALC882_FIXUP_ACER_ASPIRE_4930G),
         SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+        SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
         SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
         SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
         SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e5842fe1b1e8..c7eb4d7d05c0 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
         /* init input-src */
         for (i = 0; i < spec->num_adc_nids; i++) {
                 int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+                /* secondary ADCs must have the unique MUX */
+                if (i > 0 && !spec->mux_nids[i])
+                        break;
                 if (spec->mux_nids[adc_idx]) {
                         int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
                         snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 9f3b01bb72c8..e0a4263baa20 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
         },
         {
                 .subvendor = 0x161f,
+                .subdevice = 0x202f,
+                .name = "Gateway M520",
+                .type = AC97_TUNE_INV_EAPD
+        },
+        {
+                .subvendor = 0x161f,
                 .subdevice = 0x203a,
                 .name = "Gateway 4525GZ",               /* AD1981B */
                 .type = AC97_TUNE_INV_EAPD
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db6c89a28bda..ea4a82d01160 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
 {
         struct fsi_priv *fsi = fsi_get_priv(substream);
         struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
-        int samples_pos = io->buff_sample_pos - 1;
 
-        if (samples_pos < 0)
-                samples_pos = 0;
-
-        return fsi_sample2frame(fsi, samples_pos);
+        return fsi_sample2frame(fsi, io->buff_sample_pos);
 }
 
 static struct snd_pcm_ops fsi_pcm_ops = {
diff --git a/sound/usb/card.h b/sound/usb/card.h
index a39edcc32a93..da5fa1ac4eda 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -1,6 +1,7 @@
 #ifndef __USBAUDIO_CARD_H
 #define __USBAUDIO_CARD_H
 
+#define MAX_NR_RATES    1024
 #define MAX_PACKS       20
 #define MAX_PACKS_HS    (MAX_PACKS * 8) /* in high speed mode */
 #define MAX_URBS        8
diff --git a/sound/usb/format.c b/sound/usb/format.c
index e09aba19375c..ddfef57c4c9f 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
         return 0;
 }
 
-#define MAX_UAC2_NR_RATES 1024
-
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
                         fp->rates |= snd_pcm_rate_to_rate_bit(rate);
 
                         nr_rates++;
-                        if (nr_rates >= MAX_UAC2_NR_RATES) {
+                        if (nr_rates >= MAX_NR_RATES) {
                                 snd_printk(KERN_ERR "invalid uac2 rates\n");
                                 break;
                         }
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a3ddac0deffd..27817266867a 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
         unsigned *rate_table = NULL;
 
         fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
-        if (! fp) {
+        if (!fp) {
                 snd_printk(KERN_ERR "cannot memdup\n");
                 return -ENOMEM;
         }
+        if (fp->nr_rates > MAX_NR_RATES) {
+                kfree(fp);
+                return -EINVAL;
+        }
         if (fp->nr_rates > 0) {
                 rate_table = kmemdup(fp->rate_table,
                                      sizeof(int) * fp->nr_rates, GFP_KERNEL);