From f2ecf903ef06eb1bbbfa969db9889643d487e73a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Mar 2020 09:21:48 +0100 Subject: [PATCH 1/8] ALSA: pcm: oss: Avoid plugin buffer overflow Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 240e4702c098..c9401832967c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -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); From d683469b3c93d7e2afd39e6e1970f24700eb7a68 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Mar 2020 10:59:22 +0100 Subject: [PATCH 2/8] ALSA: line6: Fix endless MIDI read loop The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) 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 From 3b36b13d5e69d6f51ff1c55d1b404a74646c9757 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 11 Mar 2020 14:13:28 +0800 Subject: [PATCH 3/8] ALSA: hda/realtek: Fix pop noise on ALC225 Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20200311061328.17614-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ac06ff1a17c..7b83b020ac3c 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; From 5461e0530c222129dfc941058be114b5cbc00837 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2020 16:57:30 +0100 Subject: [PATCH 4/8] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/20200312155730.7520-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index c9401832967c..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); From 4384f167ce5fa7241b61bb0984d651bc528ddebe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2020 10:05:05 +0100 Subject: [PATCH 5/8] ALSA: seq: virmidi: Fix running status after receiving sysex The virmidi driver handles sysex event exceptionally in a short-cut snd_seq_dump_var_event() call, but this missed the reset of the running status. As a result, it may lead to an incomplete command right after the sysex when an event with the same running status was queued. Fix it by clearing the running status properly via alling snd_midi_event_reset_decode() for that code path. Reported-by: Andreas Steinmetz Cc: Link: https://lore.kernel.org/r/3b4a4e0f232b7afbaf0a843f63d0e538e3029bfd.camel@domdv.de Link: https://lore.kernel.org/r/20200316090506.23966-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_virmidi.c | 1 + 1 file changed, 1 insertion(+) 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) From 6c3171ef76a0bad892050f6959a7eac02fb16df7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2020 10:05:06 +0100 Subject: [PATCH 6/8] ALSA: seq: oss: Fix running status after receiving sysex This is a similar bug like the previous case for virmidi: the invalid running status is kept after receiving a sysex message. Again the fix is to clear the running status after handling the sysex. Cc: Link: https://lore.kernel.org/r/3b4a4e0f232b7afbaf0a843f63d0e538e3029bfd.camel@domdv.de Link: https://lore.kernel.org/r/20200316090506.23966-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_midi.c | 1 + 1 file changed, 1 insertion(+) 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) From d858c706bdca97698752bd26b60c21ec07ef04f2 Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Tue, 17 Mar 2020 16:28:07 +0800 Subject: [PATCH 7/8] ALSA: hda/realtek - Enable headset mic of Acer X2660G with ALC662 The Acer desktop X2660G with ALC662 can't detect the headset microphone until ALC662_FIXUP_ACER_X2660G_HEADSET_MODE quirk applied. Signed-off-by: Jian-Hong Pan Cc: Link: https://lore.kernel.org/r/20200317082806.73194-2-jian-hong@endlessm.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b83b020ac3c..a08481f358e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8612,6 +8612,7 @@ enum { ALC669_FIXUP_ACER_ASPIRE_ETHOS, ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET, ALC671_FIXUP_HP_HEADSET_MIC2, + ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, }; static const struct hda_fixup alc662_fixups[] = { @@ -8957,6 +8958,15 @@ 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 + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -8968,6 +8978,7 @@ 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, 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), From a124458a127ccd7629e20cd7bae3e1f758ed32aa Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Tue, 17 Mar 2020 16:28:09 +0800 Subject: [PATCH 8/8] ALSA: hda/realtek - Enable the headset of Acer N50-600 with ALC662 A headset on the desktop like Acer N50-600 does not work, until quirk ALC662_FIXUP_ACER_NITRO_HEADSET_MODE is applied. Signed-off-by: Jian-Hong Pan Cc: Link: https://lore.kernel.org/r/20200317082806.73194-3-jian-hong@endlessm.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a08481f358e9..63e1a56f705b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8613,6 +8613,7 @@ enum { 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[] = { @@ -8967,6 +8968,16 @@ static const struct hda_fixup alc662_fixups[] = { .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[] = { @@ -8978,6 +8989,7 @@ 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),