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-rw-r--r--sound/core/timer.c5
-rw-r--r--sound/oss/pas2_pcm.c8
-rw-r--r--sound/oss/pss.c6
-rw-r--r--sound/pci/Kconfig10
-rw-r--r--sound/pci/asihpi/hpicmn.c5
-rw-r--r--sound/pci/hda/alc269_quirks.c7
-rw-r--r--sound/pci/hda/patch_realtek.c26
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/rme9652/hdspm.c19
-rw-r--r--sound/usb/caiaq/input.c2
-rw-r--r--sound/usb/endpoint.c2
-rw-r--r--sound/usb/mixer.c25
-rw-r--r--sound/usb/mixer.h1
-rw-r--r--sound/usb/quirks-table.h6
-rw-r--r--sound/usb/quirks.c2
15 files changed, 83 insertions, 43 deletions
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0a0dc..67ebf1c21c0 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@ int snd_timer_close(struct snd_timer_instance *timeri)
mutex_unlock(&register_mutex);
} else {
timer = timeri->timer;
+ if (snd_BUG_ON(!timer))
+ goto out;
/* wait, until the active callback is finished */
spin_lock_irq(&timer->lock);
while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -353,6 +355,7 @@ int snd_timer_close(struct snd_timer_instance *timeri)
}
mutex_unlock(&register_mutex);
}
+ out:
if (timeri->private_free)
timeri->private_free(timeri);
kfree(timeri->owner);
@@ -531,6 +534,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
if (err < 0)
return err;
timer = timeri->timer;
+ if (!timer)
+ return -EINVAL;
spin_lock_irqsave(&timer->lock, flags);
timeri->cticks = timeri->ticks;
timeri->pticks = 0;
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 8f7d175767a..6f13ab4afc6 100644
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
if (pcm_channels & 2)
{
- foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
- arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+ foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
+ arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
}
else
{
- foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
- arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+ foo = (PIT_TICK_RATE + (arg / 2)) / arg;
+ arg = (PIT_TICK_RATE + (foo / 2)) / foo;
}
pcm_speed = arg;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 9b800ce5100..2fc0624024b 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@ static void configure_nonsound_components(void)
if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */
printk(KERN_INFO "PSS: CDROM port not enabled.\n");
- } else if (check_region(pss_cdrom_port, 2)) {
+ } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
+ pss_cdrom_port = -1;
printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
} else {
set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1232,7 +1233,8 @@ static void __exit cleanup_pss(void)
if(pssmpu)
unload_pss_mpu(&cfg_mpu);
unload_pss(&cfg);
- }
+ } else if (pss_cdrom_port != -1)
+ release_region(pss_cdrom_port, 2);
if(!pss_keep_settings) /* Keep hardware settings if asked */
{
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5bf8e0..88168044375 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
# ALSA PCI drivers
+config SND_TEA575X
+ tristate
+ depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+ default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
menuconfig SND_PCI
bool "PCI sound devices"
depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
SF64-PCR) into the snd-fm801 driver.
-config SND_TEA575X
- tristate
- depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
- default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
source "sound/pci/hda/Kconfig"
config SND_HDSP
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca13115..bd47521b24e 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
if (!p_cache)
return NULL;
- p_cache->p_info =
- kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+ p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+ GFP_KERNEL);
if (!p_cache->p_info) {
kfree(p_cache);
return NULL;
}
- memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
p_cache->cache_size_in_bytes = size_in_bytes;
p_cache->control_count = control_count;
p_cache->p_cache = p_dsp_control_buffer;
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf29b15..5ac0e2162a4 100644
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60fe35..9a1aa09f47f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4484,6 +4484,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
}
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ int coef;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ /* The digital-mic unit sends PDM (differential signal) instead of
+ * the standard PCM, thus you can't record a valid mono stream as is.
+ * Below is a workaround specific to ALC269 to control the dmic
+ * signal source as mono.
+ */
+ coef = alc_read_coef_idx(codec, 0x07);
+ alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4510,7 @@ enum {
ALC275_FIXUP_SONY_HWEQ,
ALC271_FIXUP_DMIC,
ALC269_FIXUP_PCM_44K,
+ ALC269_FIXUP_STEREO_DMIC,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4573,19 @@ static const struct alc_fixup alc269_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_pcm_44k,
},
+ [ALC269_FIXUP_STEREO_DMIC] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_stereo_dmic,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798bf33..4ebfbd874c9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
struct nid_path *path;
bool check_dac;
- hda_nid_t pin, dac;
+ hda_nid_t pin, dac = 0;
int err;
pin = spec->autocfg.speaker_pins[0];
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67ced90..493e3946756 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
break;
case MADIface:
freq_const = 131072000000000ULL;
+ break;
+ default:
+ snd_BUG();
+ return 0;
}
return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
switch (hdspm->io_type) {
case MADIface:
- n = 131072000000000ULL; /* 125 MHz */
- break;
+ n = 131072000000000ULL; /* 125 MHz */
+ break;
case MADI:
case AES32:
- n = 110069313433624ULL; /* 105 MHz */
- break;
+ n = 110069313433624ULL; /* 105 MHz */
+ break;
case RayDAT:
case AIO:
- n = 104857600000000ULL; /* 100 MHz */
- break;
+ n = 104857600000000ULL; /* 100 MHz */
+ break;
+ default:
+ snd_BUG();
+ return;
}
n = div_u64(n, rate);
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7a70a..a213813487b 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A };
static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4,
KEY_5, KEY_6, KEY_7 };
static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4,
- KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+ KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
static unsigned short keycode_kore[] = {
KEY_FN_F1, /* "menu" */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7c0d21ecd82..7d46e482375 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
continue;
}
if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
- ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+ ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
dev->devnum, iface_no, altno);
continue;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76e363..c04d7c71ac8 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1191,6 +1191,11 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
if (state->mixer->protocol == UAC_VERSION_1) {
csize = hdr->bControlSize;
+ if (!csize) {
+ snd_printdd(KERN_ERR "usbaudio: unit %u: "
+ "invalid bControlSize == 0\n", unitid);
+ return -EINVAL;
+ }
channels = (hdr->bLength - 7) / csize - 1;
bmaControls = hdr->bmaControls;
} else {
@@ -1934,15 +1939,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
struct mixer_build state;
int err;
const struct usbmix_ctl_map *map;
- struct usb_host_interface *hostif;
void *p;
- hostif = mixer->chip->ctrl_intf;
memset(&state, 0, sizeof(state));
state.chip = mixer->chip;
state.mixer = mixer;
- state.buffer = hostif->extra;
- state.buflen = hostif->extralen;
+ state.buffer = mixer->hostif->extra;
+ state.buflen = mixer->hostif->extralen;
/* check the mapping table */
for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1958,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
}
p = NULL;
- while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+ while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+ p, UAC_OUTPUT_TERMINAL)) != NULL) {
if (mixer->protocol == UAC_VERSION_1) {
struct uac1_output_terminal_descriptor *desc = p;
@@ -2162,17 +2166,15 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
/* create the handler for the optional status interrupt endpoint */
static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
{
- struct usb_host_interface *hostif;
struct usb_endpoint_descriptor *ep;
void *transfer_buffer;
int buffer_length;
unsigned int epnum;
- hostif = mixer->chip->ctrl_intf;
/* we need one interrupt input endpoint */
- if (get_iface_desc(hostif)->bNumEndpoints < 1)
+ if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
return 0;
- ep = get_endpoint(hostif, 0);
+ ep = get_endpoint(mixer->hostif, 0);
if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
return 0;
@@ -2202,7 +2204,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
};
struct usb_mixer_interface *mixer;
struct snd_info_entry *entry;
- struct usb_host_interface *host_iface;
int err;
strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2220,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
return -ENOMEM;
}
- host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+ switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
case UAC_VERSION_1:
default:
mixer->protocol = UAC_VERSION_1;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14dcfe8..81b2d8a32fb 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
struct usb_mixer_interface {
struct snd_usb_audio *chip;
+ struct usb_host_interface *hostif;
struct list_head list;
unsigned int ignore_ctl_error;
struct urb *urb;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index dba0b7f11c5..4d4f86552a2 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2417,6 +2417,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.idProduct = 0x1020,
},
+/* KeithMcMillen Stringport */
+{
+ USB_DEVICE(0x1f38, 0x0001),
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+
/* Miditech devices */
{
USB_DEVICE(0x4752, 0x0011),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 77762c99afb..81e07d84258 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
*/
static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
{
- int err, reg;
+ int err = 0, reg;
int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-24 12:17:54 +0000