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.
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8e06eb..0ffccc1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@
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 @@
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 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 @@
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 @@
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 @@
.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 @@
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 e5842fe..c7eb4d7 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -666,6 +666,9 @@
/* 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 9f3b01b..e0a4263 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2102,6 +2102,12 @@
},
{
.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 db6c89a..ea4a82d0 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@
{
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 a39edcc..da5fa1a 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 e09aba1..ddfef57 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@
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 @@
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 a3ddac0..2781726 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@
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);