Merge tag 'v4.4.220' into toradex_vf_4.4-next

This is the 4.4.220 stable release

19 files changed, 111 insertions, 35 deletions

diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index c6888d76ca5e..7c5d124d538c 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->next) {
 			if (plugin->dst_frames)
 				frames = plugin->dst_frames(plugin, frames);
-			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
+			if ((snd_pcm_sframes_t)frames <= 0)
 				return -ENXIO;
 			plugin = plugin->next;
 			err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
 		while (plugin->prev) {
 			if (plugin->src_frames)
 				frames = plugin->src_frames(plugin, frames);
-			if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
+			if ((snd_pcm_sframes_t)frames <= 0)
 				return -ENXIO;
 			plugin = plugin->prev;
 			err = snd_pcm_plugin_alloc(plugin, frames);
@@ -196,7 +196,9 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
 	return 0;
 }
 
-snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug,
+					  snd_pcm_uframes_t drv_frames,
+					  bool check_size)
 {
 	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
 	int stream;
@@ -209,6 +211,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		plugin = snd_pcm_plug_last(plug);
 		while (plugin && drv_frames > 0) {
+			if (check_size && drv_frames > plugin->buf_frames)
+				drv_frames = plugin->buf_frames;
 			plugin_prev = plugin->prev;
 			if (plugin->src_frames)
 				drv_frames = plugin->src_frames(plugin, drv_frames);
@@ -220,6 +224,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p
 			plugin_next = plugin->next;
 			if (plugin->dst_frames)
 				drv_frames = plugin->dst_frames(plugin, drv_frames);
+			if (check_size && drv_frames > plugin->buf_frames)
+				drv_frames = plugin->buf_frames;
 			plugin = plugin_next;
 		}
 	} else
@@ -227,7 +233,9 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p
 	return drv_frames;
 }
 
-snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
+static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug,
+					 snd_pcm_uframes_t clt_frames,
+					 bool check_size)
 {
 	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
 	snd_pcm_sframes_t frames;
@@ -248,11 +256,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc
 				if (frames < 0)
 					return frames;
 			}
+			if (check_size && frames > plugin->buf_frames)
+				frames = plugin->buf_frames;
 			plugin = plugin_next;
 		}
 	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
 		plugin = snd_pcm_plug_last(plug);
 		while (plugin) {
+			if (check_size && frames > plugin->buf_frames)
+				frames = plugin->buf_frames;
 			plugin_prev = plugin->prev;
 			if (plugin->src_frames) {
 				frames = plugin->src_frames(plugin, frames);
@@ -266,6 +278,18 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc
 	return frames;
 }
 
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug,
+					   snd_pcm_uframes_t drv_frames)
+{
+	return plug_client_size(plug, drv_frames, false);
+}
+
+snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug,
+					  snd_pcm_uframes_t clt_frames)
+{
+	return plug_slave_size(plug, clt_frames, false);
+}
+
 static int snd_pcm_plug_formats(struct snd_mask *mask, snd_pcm_format_t format)
 {
 	struct snd_mask formats = *mask;
@@ -620,7 +644,7 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st
 		src_channels = dst_channels;
 		plugin = next;
 	}
-	return snd_pcm_plug_client_size(plug, frames);
+	return plug_client_size(plug, frames, true);
 }
 
 snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size)
@@ -630,7 +654,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str
 	snd_pcm_sframes_t frames = size;
 	int err;
 
-	frames = snd_pcm_plug_slave_size(plug, frames);
+	frames = plug_slave_size(plug, frames, true);
 	if (frames < 0)
 		return frames;
 
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index 9debd1b8fd28..cdfb8f92d554 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -615,6 +615,7 @@ send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, struct seq
 		len = snd_seq_oss_timer_start(dp->timer);
 	if (ev->type == SNDRV_SEQ_EVENT_SYSEX) {
 		snd_seq_oss_readq_sysex(dp->readq, mdev->seq_device, ev);
+		snd_midi_event_reset_decode(mdev->coder);
 	} else {
 		len = snd_midi_event_decode(mdev->coder, msg, sizeof(msg), ev);
 		if (len > 0)
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 975a7c939d2f..26b478960c66 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -95,6 +95,7 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
 			if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
 				continue;
 			snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
+			snd_midi_event_reset_decode(vmidi->parser);
 		} else {
 			len = snd_midi_event_decode(vmidi->parser, msg, sizeof(msg), ev);
 			if (len > 0)
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index c397e7da0eac..7ccfb09535e1 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
+	int chs = get_amp_channels(kcontrol);
+
 	if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
-		ucontrol->value.integer.value[0] =
+		if (chs & 1)
+			ucontrol->value.integer.value[0] = beep->enabled;
+		if (chs & 2)
 			ucontrol->value.integer.value[1] = beep->enabled;
 		return 0;
 	}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 16664b07b553..825d9b27dbe1 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -876,6 +876,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
 
 	/* power-up all before initialization */
 	hda_set_power_state(codec, AC_PWRST_D0);
+	codec->core.dev.power.power_state = PMSG_ON;
 
 	snd_hda_codec_proc_new(codec);
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3e3277100f08..faf255439702 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1839,24 +1839,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context)
 {
 	struct snd_card *card = context;
 	struct azx *chip = card->private_data;
-	struct pci_dev *pci = chip->pci;
-
-	if (!fw) {
-		dev_err(card->dev, "Cannot load firmware, aborting\n");
-		goto error;
-	}
 
-	chip->fw = fw;
+	if (fw)
+		chip->fw = fw;
+	else
+		dev_err(card->dev, "Cannot load firmware, continue without patching\n");
 	if (!chip->disabled) {
 		/* continue probing */
-		if (azx_probe_continue(chip))
-			goto error;
+		azx_probe_continue(chip);
 	}
-	return; /* OK */
-
- error:
-	snd_card_free(card);
-	pci_set_drvdata(pci, NULL);
 }
 #endif
 
@@ -1982,6 +1973,17 @@ static const struct hdac_io_ops pci_hda_io_ops = {
 	.dma_free_pages = dma_free_pages,
 };
 
+/* Blacklist for skipping the whole probe:
+ * some HD-audio PCI entries are exposed without any codecs, and such devices
+ * should be ignored from the beginning.
+ */
+static const struct snd_pci_quirk driver_blacklist[] = {
+	SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0),
+	SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0),
+	SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0),
+	{}
+};
+
 static const struct hda_controller_ops pci_hda_ops = {
 	.disable_msi_reset_irq = disable_msi_reset_irq,
 	.substream_alloc_pages = substream_alloc_pages,
@@ -2001,6 +2003,11 @@ static int azx_probe(struct pci_dev *pci,
 	bool schedule_probe;
 	int err;
 
+	if (snd_pci_quirk_lookup(pci, driver_blacklist)) {
+		dev_info(&pci->dev, "Skipping the blacklisted device\n");
+		return -ENODEV;
+	}
+
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 	if (!enable[dev]) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 55bae9e6de27..76cf438aa339 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6333,6 +6333,8 @@ static int patch_alc269(struct hda_codec *codec)
 		alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
 		break;
 	case 0x10ec0225:
+		codec->power_save_node = 1;
+		/* fall through */
 	case 0x10ec0295:
 	case 0x10ec0299:
 		spec->codec_variant = ALC269_TYPE_ALC225;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 2697402b5195..41f6450a2539 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol,
 	struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
 
 	mutex_lock(&ice->gpio_mutex);
-	ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
+	ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
 	mutex_unlock(&ice->gpio_mutex);
 	return 0;
 }
@@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&ice->gpio_mutex);
 	oval = wm_get(ice, WM_ADC_MUX);
-	nval = (oval & 0xe0) | ucontrol->value.integer.value[0];
+	nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0];
 	if (nval != oval) {
 		wm_put(ice, WM_ADC_MUX, nval);
 		change = 1;
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index d55388e082e1..b070d4754745 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -1318,7 +1318,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute)
 				dai->capture_widget->name);
 		w = dai->capture_widget;
 		snd_soc_dapm_widget_for_each_source_path(w, p) {
-			if (p->connected && !p->connected(w, p->sink))
+			if (p->connected && !p->connected(w, p->source))
 				continue;
 
 			if (p->connect &&  p->source->power &&
diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c
index 3a0b3bf0af97..e9c6894cc27f 100644
--- a/sound/soc/intel/atom/sst/sst_pci.c
+++ b/sound/soc/intel/atom/sst/sst_pci.c
@@ -107,7 +107,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx)
 	dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram);
 do_release_regions:
 	pci_release_regions(pci);
-	return 0;
+	return ret;
 }
 
 /*
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 794a3499e567..0dc1ab48fceb 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -92,7 +92,7 @@
 #define JZ_AIC_I2S_STATUS_BUSY BIT(2)
 
 #define JZ_AIC_CLK_DIV_MASK 0xf
-#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_SHIFT 0
 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
 #define I2SDIV_IDV_SHIFT 8
 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b245379b4dfc..2798f4bb7fe4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -751,7 +751,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i)
 			val = max - val;
 		p->connect = !!val;
 	} else {
-		p->connect = 0;
+		/* since a virtual mixer has no backing registers to
+		 * decide which path to connect, it will try to match
+		 * with initial state.  This is to ensure
+		 * that the default mixer choice will be
+		 * correctly powered up during initialization.
+		 */
+		p->connect = invert;
 	}
 }
 
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 2f67ba6d7a8f..acacbce2a821 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -837,7 +837,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
 	unsigned int regbase = mc->regbase;
 	unsigned int regcount = mc->regcount;
 	unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
-	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int regwmask = (1UL<<regwshift)-1;
 	unsigned int invert = mc->invert;
 	unsigned long mask = (1UL<<mc->nbits)-1;
 	long min = mc->min;
@@ -886,7 +886,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
 	unsigned int regbase = mc->regbase;
 	unsigned int regcount = mc->regcount;
 	unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
-	unsigned int regwmask = (1<<regwshift)-1;
+	unsigned int regwmask = (1UL<<regwshift)-1;
 	unsigned int invert = mc->invert;
 	unsigned long mask = (1UL<<mc->nbits)-1;
 	long max = mc->max;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 7cffa98ec313..d4bf3dc6b015 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1951,7 +1951,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 		switch (cmd) {
 		case SNDRV_PCM_TRIGGER_START:
 			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
-			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
 				continue;
 
 			ret = dpcm_do_trigger(dpcm, be_substream, cmd);
@@ -1981,7 +1982,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 			be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
 			break;
 		case SNDRV_PCM_TRIGGER_STOP:
-			if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+			if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) &&
+			    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
 				continue;
 
 			if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 824f4d7fc41f..0675ab3fec6c 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -378,7 +378,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg,
 	struct snd_soc_component *comp = tplg->comp;
 
 	return soc_tplg_add_dcontrol(comp->card->snd_card,
-				comp->dev, k, NULL, comp, kcontrol);
+				comp->dev, k, comp->name_prefix, comp, kcontrol);
 }
 
 /* remove a mixer kcontrol */
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 954dc4423cb0..ae2c35918002 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -283,7 +283,7 @@ static void line6_data_received(struct urb *urb)
 		    line6_midibuf_read(mb, line6->buffer_message,
 				       LINE6_MESSAGE_MAXLEN);
 
-		if (done == 0)
+		if (done <= 0)
 			break;
 
 		line6->message_length = done;
diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c
index 36a610ba342e..c931d48801eb 100644
--- a/sound/usb/line6/midibuf.c
+++ b/sound/usb/line6/midibuf.c
@@ -163,7 +163,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data,
 			int midi_length_prev =
 			    midibuf_message_length(this->command_prev);
 
-			if (midi_length_prev > 0) {
+			if (midi_length_prev > 1) {
 				midi_length = midi_length_prev - 1;
 				repeat = 1;
 			} else
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 73149b9be29c..f191f4a3cf3b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2269,7 +2269,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 		if (map->id == state.chip->usb_id) {
 			state.map = map->map;
 			state.selector_map = map->selector_map;
-			mixer->ignore_ctl_error = map->ignore_ctl_error;
+			mixer->ignore_ctl_error |= map->ignore_ctl_error;
 			break;
 		}
 	}
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index f5cf23ffb35b..a9f36f53d9d3 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -361,6 +361,14 @@ static const struct usbmix_name_map dell_alc4020_map[] = {
 	{ 0 }
 };
 
+/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX
+ * response for Input Gain Pad (id=19, control=12).  Skip it.
+ */
+static const struct usbmix_name_map asus_rog_map[] = {
+	{ 19, NULL, 12 }, /* FU, Input Gain Pad */
+	{}
+};
+
 /*
  * Control map entries
  */
@@ -480,6 +488,26 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x05a7, 0x1020),
 		.map = bose_companion5_map,
 	},
+	{	/* Gigabyte TRX40 Aorus Pro WiFi */
+		.id = USB_ID(0x0414, 0xa002),
+		.map = asus_rog_map,
+	},
+	{	/* ASUS ROG Zenith II */
+		.id = USB_ID(0x0b05, 0x1916),
+		.map = asus_rog_map,
+	},
+	{	/* ASUS ROG Strix */
+		.id = USB_ID(0x0b05, 0x1917),
+		.map = asus_rog_map,
+	},
+	{	/* MSI TRX40 Creator */
+		.id = USB_ID(0x0db0, 0x0d64),
+		.map = asus_rog_map,
+	},
+	{	/* MSI TRX40 */
+		.id = USB_ID(0x0db0, 0x543d),
+		.map = asus_rog_map,
+	},
 	{ 0 } /* terminator */
 };