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| author | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-09-30 11:08:44 +0000 |
|---|---|---|
| committer | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-09-30 11:08:44 +0000 |
| commit | 38c121c484e12f677c2cb6afb882cd024bd469c1 (patch) | |
| tree | 167505d54fcc2f34616b3b61f26ae560b6f4c382 /resources | |
| parent | 1795c358fcbad63ca0dea746dd74d073ae1faa22 (diff) | |
| download | webrtc-38c121c484e12f677c2cb6afb882cd024bd469c1.tar.gz | |
Minor modifications to test::RtpFileReader
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.
Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'resources')
| -rw-r--r-- | resources/video_coding/pltype103_header_only.rtp.sha1 | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/resources/video_coding/pltype103_header_only.rtp.sha1 b/resources/video_coding/pltype103_header_only.rtp.sha1 new file mode 100644 index 0000000000..eee5aa9f7b --- /dev/null +++ b/resources/video_coding/pltype103_header_only.rtp.sha1 @@ -0,0 +1 @@ +2a77f0d030aa96f51f0a96e971b42c3b11fe006b
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