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From 4cc8d6505ab82db3357613d36e6c58a297f57f7c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 4 Dec 2019 15:48:24 +0100
Subject: ALSA: pcm: oss: Avoid potential buffer overflows
From: Takashi Iwai <tiwai@suse.de>
commit 4cc8d6505ab82db3357613d36e6c58a297f57f7c upstream.
syzkaller reported an invalid access in PCM OSS read, and this seems
to be an overflow of the internal buffer allocated for a plugin.
Since the rate plugin adjusts its transfer size dynamically, the
calculation for the chained plugin might be bigger than the given
buffer size in some extreme cases, which lead to such an buffer
overflow as caught by KASAN.
Fix it by limiting the max transfer size properly by checking against
the destination size in each plugin transfer callback.
Reported-by: syzbot+f153bde47a62e0b05f83@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191204144824.17801-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
---
sound/core/oss/linear.c | 2 ++
sound/core/oss/mulaw.c | 2 ++
sound/core/oss/route.c | 2 ++
3 files changed, 6 insertions(+)
--- a/sound/core/oss/linear.c
+++ b/sound/core/oss/linear.c
@@ -107,6 +107,8 @@ static snd_pcm_sframes_t linear_transfer
}
}
#endif
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
convert(plugin, src_channels, dst_channels, frames);
return frames;
}
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -269,6 +269,8 @@ static snd_pcm_sframes_t mulaw_transfer(
}
}
#endif
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
data = (struct mulaw_priv *)plugin->extra_data;
data->func(plugin, src_channels, dst_channels, frames);
return frames;
--- a/sound/core/oss/route.c
+++ b/sound/core/oss/route.c
@@ -57,6 +57,8 @@ static snd_pcm_sframes_t route_transfer(
return -ENXIO;
if (frames == 0)
return 0;
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
nsrcs = plugin->src_format.channels;
ndsts = plugin->dst_format.channels;
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