[FFmpeg-devel] [PATCH 1/5] Use av_sat_add64() when updating start_time by skip_samples.

[Dale Curtis]

Avoids overflow from fuzzed skip_samples values.

Signed-off-by: Dale Curtis <dalecur...@chromium.org>
---
 libavformat/utils.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

From 7482aaef44fa4c6c43efd16b2ed8eb474b1283b0 Mon Sep 17 00:00:00 2001
From: Dale Curtis <dalecurtis@chromium.org>
Date: Thu, 14 May 2020 14:29:15 -0700
Subject: [PATCH 1/5] Use av_sat_add64() when updating start_time by
 skip_samples.

Avoids overflow from fuzzed skip_samples values.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
---
 libavformat/utils.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavformat/utils.c b/libavformat/utils.c
index 62c70fb9d6..107ab05b9a 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -1156,7 +1156,7 @@ static void update_initial_timestamps(AVFormatContext *s, int stream_index,
         if (st->start_time == AV_NOPTS_VALUE && pktl_it->pkt.pts != AV_NOPTS_VALUE) {
             st->start_time = pktl_it->pkt.pts;
             if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->sample_rate)
-                st->start_time += av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base);
+                st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base));
         }
     }
 
@@ -1169,7 +1169,7 @@ static void update_initial_timestamps(AVFormatContext *s, int stream_index,
             st->start_time = pts;
         }
         if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->sample_rate)
-            st->start_time += av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base);
+            st->start_time = av_sat_add64(st->start_time, av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate}, st->time_base));
     }
 }
 
-- 
2.26.2.761.g0e0b3e54be-goog


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