[FFmpeg-cvslog] avformat/dsfdec: properly handle padded last packet

[Paul B Mahol]

ffmpeg | branch: release/4.0 | Paul B Mahol <onemda at gmail.com> | Sun Apr 22 11:52:32 2018 +0200| [e44498984d3d1b52640c597d88d534b07d2e7881] | committer: Paul B Mahol

avformat/dsfdec: properly handle padded last packet

Fixes #5489.

Signed-off-by: Paul B Mahol <onemda at gmail.com>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e44498984d3d1b52640c597d88d534b07d2e7881
---

 libavformat/dsfdec.c | 36 ++++++++++++++++++++++++++++++++++--
 1 file changed, 34 insertions(+), 2 deletions(-)

diff --git a/libavformat/dsfdec.c b/libavformat/dsfdec.c
index 41538fd83c..e0485f1492 100644
--- a/libavformat/dsfdec.c
+++ b/libavformat/dsfdec.c
@@ -26,6 +26,8 @@
 
 typedef struct {
     uint64_t data_end;
+    uint64_t audio_size;
+    uint64_t data_size;
 } DSFContext;
 
 static int dsf_probe(AVProbeData *p)
@@ -120,7 +122,7 @@ static int dsf_read_header(AVFormatContext *s)
         return AVERROR_INVALIDDATA;
     }
 
-    avio_skip(pb, 8);
+    dsf->audio_size = avio_rl64(pb) / 8 * st->codecpar->channels;
     st->codecpar->block_align = avio_rl32(pb);
     if (st->codecpar->block_align > INT_MAX / st->codecpar->channels) {
         avpriv_request_sample(s, "block_align overflow");
@@ -135,7 +137,9 @@ static int dsf_read_header(AVFormatContext *s)
     dsf->data_end = avio_tell(pb);
     if (avio_rl32(pb) != MKTAG('d', 'a', 't', 'a'))
         return AVERROR_INVALIDDATA;
-    dsf->data_end += avio_rl64(pb);
+    dsf->data_size = avio_rl64(pb);
+    dsf->data_end += dsf->data_size;
+    s->internal->data_offset = avio_tell(pb);
 
     return 0;
 }
@@ -151,6 +155,34 @@ static int dsf_read_packet(AVFormatContext *s, AVPacket *pkt)
         return AVERROR_EOF;
 
     pkt->stream_index = 0;
+    if (dsf->data_size > dsf->audio_size) {
+        int last_packet = pos == (dsf->data_end - st->codecpar->block_align);
+
+        if (last_packet) {
+            int64_t data_pos = pos - s->internal->data_offset;
+            int64_t packet_size = dsf->audio_size - data_pos;
+            int64_t skip_size = dsf->data_size - data_pos - packet_size;
+            uint8_t *dst;
+            int ch, ret;
+
+            if (packet_size <= 0 || skip_size <= 0)
+                return AVERROR_INVALIDDATA;
+
+            if (av_new_packet(pkt, packet_size) < 0)
+                return AVERROR(ENOMEM);
+            dst = pkt->data;
+            for (ch = 0; ch < st->codecpar->channels; ch++) {
+                ret = avio_read(pb, dst,  packet_size / st->codecpar->channels);
+                if (ret < packet_size / st->codecpar->channels)
+                    return AVERROR_EOF;
+
+                dst += ret;
+                avio_skip(pb, skip_size / st->codecpar->channels);
+            }
+
+            return 0;
+        }
+    }
     return av_get_packet(pb, pkt, FFMIN(dsf->data_end - pos, st->codecpar->block_align));
 }