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-rw-r--r--sound/pci/hda/patch_realtek.c42
-rw-r--r--sound/pci/hda/patch_sigmatel.c4
-rw-r--r--sound/pci/hda/patch_via.c3
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/soc/codecs/ak4642.c31
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/sh/fsi.c6
-rw-r--r--sound/usb/caiaq/audio.c5
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/format.c4
-rw-r--r--sound/usb/quirks.c6
11 files changed, 79 insertions, 31 deletions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9350f3c3bdf8..3647baa9bfed 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;
@@ -3149,7 +3151,10 @@ 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;
}
@@ -4374,6 +4379,7 @@ 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,
@@ -4427,6 +4433,31 @@ 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) {
+ /* fake the connections during parsing the tree */
+ 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);
+ } else if (action == ALC_FIXUP_ACT_PROBE) {
+ /* restore the connections */
+ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+ snd_hda_override_conn_list(codec, 0x14, 5, conn);
+ snd_hda_override_conn_list(codec, 0x15, 5, conn);
+ snd_hda_override_conn_list(codec, 0x18, 5, conn);
+ snd_hda_override_conn_list(codec, 0x1a, 5, conn);
+ }
+}
+
static const struct alc_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = ALC_FIXUP_PINS,
@@ -4574,6 +4605,10 @@ 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[] = {
@@ -4598,6 +4633,7 @@ 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_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 948f0be2f4f3..6345df131a00 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
spec->gpio_dir, spec->gpio_data);
} else {
notmtd_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD;
+ AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
muted_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
+ AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50;
spec->vref_led = muted ? muted_lvl : notmtd_lvl;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 284e311040fe..dff9a00ee8fb 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/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..278c0a0575f5 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 29c4b02c4790..0ac228b7dc04 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
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/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed4..fde9a7a29cb6 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
spin_lock(&dev->spinlock);
- if (dev->input_panic || dev->output_panic)
+ if (dev->input_panic || dev->output_panic) {
ptr = SNDRV_PCM_POS_XRUN;
+ goto unlock;
+ }
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
ptr = bytes_to_frames(sub->runtime,
dev->audio_in_buf_pos[index]);
+unlock:
spin_unlock(&dev->spinlock);
return ptr;
}
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);
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2024-07-16 10:36:50 +0200