Merge tag 'sound-5.6-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A few fixes covering the issues reported by syzkaller, a couple of
  fixes for the MIDI decoding bug, and a few usual HD-audio quirks.

  Some of them are about ALSA core stuff, but they are small fixes just
  for corner cases, and nothing thrilling"

* tag 'sound-5.6-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek - Enable the headset of Acer N50-600 with ALC662
  ALSA: hda/realtek - Enable headset mic of Acer X2660G with ALC662
  ALSA: seq: oss: Fix running status after receiving sysex
  ALSA: seq: virmidi: Fix running status after receiving sysex
  ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
  ALSA: hda/realtek: Fix pop noise on ALC225
  ALSA: line6: Fix endless MIDI read loop
  ALSA: pcm: oss: Avoid plugin buffer overflow

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);
@@ -209,6 +209,8 @@ 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 (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 +222,8 @@ 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 (drv_frames > plugin->buf_frames)
+				drv_frames = plugin->buf_frames;
 			plugin = plugin_next;
 		}
 	} else
@@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc
 				if (frames < 0)
 					return frames;
 			}
+			if (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 (frames > plugin->buf_frames)
+				frames = plugin->buf_frames;
 			plugin_prev = plugin->prev;
 			if (plugin->src_frames) {
 				frames = plugin->src_frames(plugin, frames);

sound/core/seq/oss/seq_oss_midi.c

@@ -602,6 +602,7 @@ 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)

sound/core/seq/seq_virmidi.c

@@ -81,6 +81,7 @@ 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)

sound/pci/hda/patch_realtek.c

@@ -8051,6 +8051,8 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->gen.mixer_nid = 0;
 		break;
 	case 0x10ec0225:
+		codec->power_save_node = 1;
+		/* fall through */
 	case 0x10ec0295:
 	case 0x10ec0299:
 		spec->codec_variant = ALC269_TYPE_ALC225;
@@ -8610,6 +8612,8 @@ enum {
 	ALC669_FIXUP_ACER_ASPIRE_ETHOS,
 	ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET,
 	ALC671_FIXUP_HP_HEADSET_MIC2,
+	ALC662_FIXUP_ACER_X2660G_HEADSET_MODE,
+	ALC662_FIXUP_ACER_NITRO_HEADSET_MODE,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -8955,6 +8959,25 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc671_fixup_hp_headset_mic2,
 	},
+	[ALC662_FIXUP_ACER_X2660G_HEADSET_MODE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02a1113c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_USI_FUNC
+	},
+	[ALC662_FIXUP_ACER_NITRO_HEADSET_MODE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */
+			{ 0x1b, 0x0221144f },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC662_FIXUP_USI_FUNC
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -8966,6 +8989,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x123c, "Acer Nitro N50-600", ALC662_FIXUP_ACER_NITRO_HEADSET_MODE),
+	SND_PCI_QUIRK(0x1025, 0x124e, "Acer 2660G", ALC662_FIXUP_ACER_X2660G_HEADSET_MODE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13),

sound/usb/line6/driver.c

@@ -305,7 +305,7 @@ static void line6_data_received(struct urb *urb)
 				line6_midibuf_read(mb, line6->buffer_message,
 						LINE6_MIDI_MESSAGE_MAXLEN);
 
-			if (done == 0)
+			if (done <= 0)
 				break;
 
 			line6->message_length = done;

sound/usb/line6/midibuf.c

@@ -159,7 +159,7 @@ 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