[GIT PULL] sound fixes for 5.6-rc7
From: Takashi Iwai
Date: Fri Mar 20 2020 - 07:12:18 EST
Linus,
please pull sound fixes for v5.6-rc7 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-5.6-rc7
The topmost commit is a124458a127ccd7629e20cd7bae3e1f758ed32aa
----------------------------------------------------------------
sound fixes for 5.6-rc7
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.
----------------------------------------------------------------
Jian-Hong Pan (2):
ALSA: hda/realtek - Enable headset mic of Acer X2660G with ALC662
ALSA: hda/realtek - Enable the headset of Acer N50-600 with ALC662
Kai-Heng Feng (1):
ALSA: hda/realtek: Fix pop noise on ALC225
Takashi Iwai (5):
ALSA: pcm: oss: Avoid plugin buffer overflow
ALSA: line6: Fix endless MIDI read loop
ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
ALSA: seq: virmidi: Fix running status after receiving sysex
ALSA: seq: oss: Fix running status after receiving sysex
---
sound/core/oss/pcm_plugin.c | 12 ++++++++++--
sound/core/seq/oss/seq_oss_midi.c | 1 +
sound/core/seq/seq_virmidi.c | 1 +
sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++++++++++
sound/usb/line6/driver.c | 2 +-
sound/usb/line6/midibuf.c | 2 +-
6 files changed, 39 insertions(+), 4 deletions(-)
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 240e4702c098..752d078908e9 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);
@@ -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);
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index a88c235b2ea3..2ddfe2226651 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/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)
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 626d87c1539b..77d7037d1476 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/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)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ac06ff1a17c..63e1a56f705b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/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),
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index b5a3f754a4f1..4f096685ed65 100644
--- a/sound/usb/line6/driver.c
+++ b/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;
diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c
index 8d6eefa0d936..6a70463f82c4 100644
--- a/sound/usb/line6/midibuf.c
+++ b/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