---

yaml
---
r: 262958
b: refs/heads/master
c: c012cdc
h: refs/heads/master
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: f6b864a9071e21186476910613ec9913b56067a2
+refs/heads/master: c012cdc858db36ff2314469930bfcfb00eef6f29

trunk/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;

trunk/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;

trunk/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 */
 	{

trunk/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

trunk/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;

trunk/sound/pci/azt3328.c

@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS)
-		return -ENODEV;
+	if (dev >= SNDRV_CARDS) {
+		err = -ENODEV;
+		goto out;
+	}
 	if (!enable[dev]) {
 		dev++;
-		return -ENOENT;
+		err = -ENOENT;
+		goto out;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		return err;
+		goto out;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");

trunk/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),

trunk/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),

trunk/sound/pci/hda/patch_sigmatel.c

@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
 	return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)

trunk/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];

trunk/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);

trunk/sound/usb/caiaq/audio.c

@@ -614,6 +614,7 @@ static void read_completed(struct urb *urb)
 	struct snd_usb_caiaqdev *dev;
 	struct urb *out;
 	int frame, len, send_it = 0, outframe = 0;
+	size_t offset = 0;
 
 	if (urb->status || !info)
 		return;
@@ -634,7 +635,8 @@ static void read_completed(struct urb *urb)
 		len = urb->iso_frame_desc[outframe].actual_length;
 		out->iso_frame_desc[outframe].length = len;
 		out->iso_frame_desc[outframe].actual_length = 0;
-		out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+		out->iso_frame_desc[outframe].offset = offset;
+		offset += len;
 
 		if (len > 0) {
 			spin_lock(&dev->spinlock);
@@ -650,7 +652,7 @@ static void read_completed(struct urb *urb)
 	}
 
 	if (send_it) {
-		out->number_of_packets = FRAMES_PER_URB;
+		out->number_of_packets = outframe;
 		out->transfer_flags = URB_ISO_ASAP;
 		usb_submit_urb(out, GFP_ATOMIC);
 	}

trunk/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"               */

trunk/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;

trunk/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;

trunk/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;