Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/tools
Henrik Lundin b0b54259c3 Let rtp_analyze parse absolute sender time
Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51559004

Cr-Commit-Position: refs/heads/master@{#9025}
2015-04-17 09:46:56 +00:00
..
2014-06-09 08:10:28 +00:00