Linux-libre 4.5.1-gnupck-4.5.1-gnu

14 files changed, 145 insertions, 46 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6b5a811e0..3a9b66c6e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -322,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			pcm_err(substream->pcm,
-				"BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
+				"invalid position: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
 				name, pos, runtime->buffer_size,
 				runtime->period_size);
 		}
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index e361024ea..d1a4d6973 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -611,6 +611,22 @@ int snd_hdac_power_up_pm(struct hdac_device *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm);
 
+/* like snd_hdac_power_up_pm(), but only increment the pm count when
+ * already powered up.  Returns -1 if not powered up, 1 if incremented
+ * or 0 if unchanged.  Only used in hdac_regmap.c
+ */
+int snd_hdac_keep_power_up(struct hdac_device *codec)
+{
+	if (!atomic_inc_not_zero(&codec->in_pm)) {
+		int ret = pm_runtime_get_if_in_use(&codec->dev);
+		if (!ret)
+			return -1;
+		if (ret < 0)
+			return 0;
+	}
+	return 1;
+}
+
 /**
  * snd_hdac_power_down_pm - power down the codec
  * @codec: the codec object
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index eb8f7c30c..bdbcd6b75 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -21,13 +21,16 @@
 #include <sound/hdaudio.h>
 #include <sound/hda_regmap.h>
 
-#ifdef CONFIG_PM
-#define codec_is_running(codec)				\
-	(atomic_read(&(codec)->in_pm) ||		\
-	 !pm_runtime_suspended(&(codec)->dev))
-#else
-#define codec_is_running(codec)		true
-#endif
+static int codec_pm_lock(struct hdac_device *codec)
+{
+	return snd_hdac_keep_power_up(codec);
+}
+
+static void codec_pm_unlock(struct hdac_device *codec, int lock)
+{
+	if (lock == 1)
+		snd_hdac_power_down_pm(codec);
+}
 
 #define get_verb(reg)	(((reg) >> 8) & 0xfff)
 
@@ -238,20 +241,28 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val)
 	struct hdac_device *codec = context;
 	int verb = get_verb(reg);
 	int err;
+	int pm_lock = 0;
 
-	if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE)
-		return -EAGAIN;
+	if (verb != AC_VERB_GET_POWER_STATE) {
+		pm_lock = codec_pm_lock(codec);
+		if (pm_lock < 0)
+			return -EAGAIN;
+	}
 	reg |= (codec->addr << 28);
-	if (is_stereo_amp_verb(reg))
-		return hda_reg_read_stereo_amp(codec, reg, val);
-	if (verb == AC_VERB_GET_PROC_COEF)
-		return hda_reg_read_coef(codec, reg, val);
+	if (is_stereo_amp_verb(reg)) {
+		err = hda_reg_read_stereo_amp(codec, reg, val);
+		goto out;
+	}
+	if (verb == AC_VERB_GET_PROC_COEF) {
+		err = hda_reg_read_coef(codec, reg, val);
+		goto out;
+	}
 	if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE)
 		reg &= ~AC_AMP_FAKE_MUTE;
 
 	err = snd_hdac_exec_verb(codec, reg, 0, val);
 	if (err < 0)
-		return err;
+		goto out;
 	/* special handling for asymmetric reads */
 	if (verb == AC_VERB_GET_POWER_STATE) {
 		if (*val & AC_PWRST_ERROR)
@@ -259,7 +270,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val)
 		else /* take only the actual state */
 			*val = (*val >> 4) & 0x0f;
 	}
-	return 0;
+ out:
+	codec_pm_unlock(codec, pm_lock);
+	return err;
 }
 
 static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
@@ -267,6 +280,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
 	struct hdac_device *codec = context;
 	unsigned int verb;
 	int i, bytes, err;
+	int pm_lock = 0;
 
 	if (codec->caps_overwriting)
 		return 0;
@@ -275,14 +289,21 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
 	reg |= (codec->addr << 28);
 	verb = get_verb(reg);
 
-	if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE)
-		return codec->lazy_cache ? 0 : -EAGAIN;
+	if (verb != AC_VERB_SET_POWER_STATE) {
+		pm_lock = codec_pm_lock(codec);
+		if (pm_lock < 0)
+			return codec->lazy_cache ? 0 : -EAGAIN;
+	}
 
-	if (is_stereo_amp_verb(reg))
-		return hda_reg_write_stereo_amp(codec, reg, val);
+	if (is_stereo_amp_verb(reg)) {
+		err = hda_reg_write_stereo_amp(codec, reg, val);
+		goto out;
+	}
 
-	if (verb == AC_VERB_SET_PROC_COEF)
-		return hda_reg_write_coef(codec, reg, val);
+	if (verb == AC_VERB_SET_PROC_COEF) {
+		err = hda_reg_write_coef(codec, reg, val);
+		goto out;
+	}
 
 	switch (verb & 0xf00) {
 	case AC_VERB_SET_AMP_GAIN_MUTE:
@@ -319,10 +340,12 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val)
 		reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff);
 		err = snd_hdac_exec_verb(codec, reg, 0, NULL);
 		if (err < 0)
-			return err;
+			goto out;
 	}
 
-	return 0;
+ out:
+	codec_pm_unlock(codec, pm_lock);
+	return err;
 }
 
 static const struct regmap_config hda_regmap_cfg = {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c1c855a6c..a47e8ae0e 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec)
 	snd_hda_gen_update_outputs(codec);
 
 	if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) {
-		spec->gpio_data = spec->gen.hp_jack_present ?
-			spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+		if (spec->gen.automute_speaker)
+			spec->gpio_data = spec->gen.hp_jack_present ?
+				spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+		else
+			spec->gpio_data =
+				spec->gpio_eapd_hp | spec->gpio_eapd_speaker;
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, spec->gpio_data);
 	}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6122b8ca8..56fefbd85 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -204,8 +204,13 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	if (codec->core.vendor_id != 0x14f150f2)
+	switch (codec->core.vendor_id) {
+	case 0x14f150f2: /* CX20722 */
+	case 0x14f150f4: /* CX20724 */
+		break;
+	default:
 		return;
+	}
 
 	/* Turn the CX20722 codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bcbc4ee10..e68fa449e 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -152,13 +152,17 @@ struct hdmi_spec {
 	struct hda_pcm_stream pcm_playback;
 
 	/* i915/powerwell (Haswell+/Valleyview+) specific */
+	bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */
 	struct i915_audio_component_audio_ops i915_audio_ops;
 	bool i915_bound; /* was i915 bound in this driver? */
 };
 
 #ifdef CONFIG_SND_HDA_I915
-#define codec_has_acomp(codec) \
-	((codec)->bus->core.audio_component != NULL)
+static inline bool codec_has_acomp(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	return spec->use_acomp_notifier;
+}
 #else
 #define codec_has_acomp(codec)	false
 #endif
@@ -1562,6 +1566,7 @@ static void update_eld(struct hda_codec *codec,
 			   eld->eld_size) != 0)
 			eld_changed = true;
 
+	pin_eld->monitor_present = eld->monitor_present;
 	pin_eld->eld_valid = eld->eld_valid;
 	pin_eld->eld_size = eld->eld_size;
 	if (eld->eld_valid)
@@ -1665,11 +1670,10 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 	int size;
 
 	mutex_lock(&per_pin->lock);
+	eld->monitor_present = false;
 	size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid,
 				      &eld->monitor_present, eld->eld_buffer,
 				      ELD_MAX_SIZE);
-	if (size < 0)
-		goto unlock;
 	if (size > 0) {
 		size = min(size, ELD_MAX_SIZE);
 		if (snd_hdmi_parse_eld(codec, &eld->info,
@@ -1873,7 +1877,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	/* Call sync_audio_rate to set the N/CTS/M manually if necessary */
 	/* Todo: add DP1.2 MST audio support later */
-	snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
+	if (codec_has_acomp(codec))
+		snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate);
 
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
 	mutex_lock(&per_pin->lock);
@@ -2432,6 +2437,10 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
 	struct hda_codec *codec = audio_ptr;
 	int pin_nid = port + 0x04;
 
+	/* we assume only from port-B to port-D */
+	if (port < 1 || port > 3)
+		return;
+
 	/* skip notification during system suspend (but not in runtime PM);
 	 * the state will be updated at resume
 	 */
@@ -2456,11 +2465,24 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 	codec->spec = spec;
 	hdmi_array_init(spec, 4);
 
-	/* Try to bind with i915 for any Intel codecs (if not done yet) */
-	if (!codec_has_acomp(codec) &&
-	    (codec->core.vendor_id >> 16) == 0x8086)
-		if (!snd_hdac_i915_init(&codec->bus->core))
-			spec->i915_bound = true;
+#ifdef CONFIG_SND_HDA_I915
+	/* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */
+	if ((codec->core.vendor_id >> 16) == 0x8086 &&
+	    is_haswell_plus(codec)) {
+#if 0
+		/* on-demand binding leads to an unbalanced refcount when
+		 * both i915 and hda drivers are probed concurrently;
+		 * disabled temporarily for now
+		 */
+		if (!codec->bus->core.audio_component)
+			if (!snd_hdac_i915_init(&codec->bus->core))
+				spec->i915_bound = true;
+#endif
+		/* use i915 audio component notifier for hotplug */
+		if (codec->bus->core.audio_component)
+			spec->use_acomp_notifier = true;
+	}
+#endif
 
 	if (is_haswell_plus(codec)) {
 		intel_haswell_enable_all_pins(codec, true);
@@ -3659,6 +3681,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",	patch_via_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 93d2156b6..4f5ca0b9c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5556,6 +5556,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
+	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 42bcbac80..ccdab29a8 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2879,6 +2879,7 @@ static void intel8x0_measure_ac97_clock(struct intel8x0 *chip)
 
 static struct snd_pci_quirk intel8x0_clock_list[] = {
 	SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
+	SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000),
 	SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
 	SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
 	SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 2ed260b10..7ccbcaf6a 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -285,6 +285,8 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
 	unsigned char data[3];
 	int err, crate;
 
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	/* if endpoint doesn't have sampling rate control, bail out */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7b1cb365f..c07a7eda4 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -438,6 +438,9 @@ exit_clear:
  *
  * New endpoints will be added to chip->ep_list and must be freed by
  * calling snd_usb_endpoint_free().
+ *
+ * For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that
+ * bNumEndpoints > 1 beforehand.
  */
 struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 					      struct usb_host_interface *alts,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 279025650..f6c3bf79a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1519,7 +1519,11 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol,
 
 	/* use known values for that card: interface#1 altsetting#1 */
 	iface = usb_ifnum_to_if(chip->dev, 1);
+	if (!iface || iface->num_altsetting < 2)
+		return -EINVAL;
 	alts = &iface->altsetting[1];
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	err = snd_usb_ctl_msg(chip->dev,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 9245f52d4..44d178ee9 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -159,6 +159,8 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 	unsigned char data[1];
 	int err;
 
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	data[0] = 1;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index c458d60d5..cd7eac28e 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 		usb_audio_err(chip, "cannot memdup\n");
 		return -ENOMEM;
 	}
+	INIT_LIST_HEAD(&fp->list);
 	if (fp->nr_rates > MAX_NR_RATES) {
 		kfree(fp);
 		return -EINVAL;
@@ -167,19 +168,20 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	stream = (fp->endpoint & USB_DIR_IN)
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 	err = snd_usb_add_audio_stream(chip, stream, fp);
-	if (err < 0) {
-		kfree(fp);
-		kfree(rate_table);
-		return err;
-	}
+	if (err < 0)
+		goto error;
 	if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
 	    fp->altset_idx >= iface->num_altsetting) {
-		kfree(fp);
-		kfree(rate_table);
-		return -EINVAL;
+		err = -EINVAL;
+		goto error;
 	}
 	alts = &iface->altsetting[fp->altset_idx];
 	altsd = get_iface_desc(alts);
+	if (altsd->bNumEndpoints < 1) {
+		err = -EINVAL;
+		goto error;
+	}
+
 	fp->protocol = altsd->bInterfaceProtocol;
 
 	if (fp->datainterval == 0)
@@ -190,6 +192,12 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	snd_usb_init_pitch(chip, fp->iface, alts, fp);
 	snd_usb_init_sample_rate(chip, fp->iface, alts, fp, fp->rate_max);
 	return 0;
+
+ error:
+	list_del(&fp->list); /* unlink for avoiding double-free */
+	kfree(fp);
+	kfree(rate_table);
+	return err;
 }
 
 static int create_auto_pcm_quirk(struct snd_usb_audio *chip,
@@ -462,6 +470,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 	fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
 	fp->datainterval = 0;
 	fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+	INIT_LIST_HEAD(&fp->list);
 
 	switch (fp->maxpacksize) {
 	case 0x120:
@@ -485,6 +494,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 	err = snd_usb_add_audio_stream(chip, stream, fp);
 	if (err < 0) {
+		list_del(&fp->list); /* unlink for avoiding double-free */
 		kfree(fp);
 		return err;
 	}
@@ -1121,6 +1131,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	switch (chip->usb_id) {
 	case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema  */
 	case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */
+	case USB_ID(0x045E, 0x076E): /* MS Lifecam HD-5001 */
 	case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */
 	case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */
 	case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index c4dc577ab..8e9548bc1 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -314,7 +314,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
 /*
  * add this endpoint to the chip instance.
  * if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
+ * if not, create a new pcm stream. note, fp is added to the substream
+ * fmt_list and will be freed on the chip instance release. do not free
+ * fp or do remove it from the substream fmt_list to avoid double-free.
  */
 int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 			     int stream,
@@ -675,6 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 					* (fp->maxpacksize & 0x7ff);
 		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
 		fp->clock = clock;
+		INIT_LIST_HEAD(&fp->list);
 
 		/* some quirks for attributes here */
 
@@ -723,6 +726,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 		dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
 		err = snd_usb_add_audio_stream(chip, stream, fp);
 		if (err < 0) {
+			list_del(&fp->list); /* unlink for avoiding double-free */
 			kfree(fp->rate_table);
 			kfree(fp->chmap);
 			kfree(fp);