be4959535a
Reason for revert: Breaks Chromium WebRTC FYI bots. Updating projects from gyp files... gyp: /b/build/slave/linux/build/src/third_party/gflags/gflags.gyp not found (cwd: /b/build/slave/linux/build) Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/linux/build Original issue's description: > Tool to convert RtcEventLog files to RtpDump format. > > This is a small utility that reads RtcEventLog files, and converts the RTP headers within it to RtpDump format. All other types of events are ignored. Three command-line flags are supported, --audio-only, --video-only and --data-only. When one of these flags is supplied, only RTP packets that match the requested type are converted. > > BUG=webrtc:4741 > R=henrik.lundin@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org, terelius@webrtc.org > > Committed: https://crrev.com/35624c2c3686a2ad40daffe073aa78507b0ef88e > Cr-Commit-Position: refs/heads/master@{#9980} TBR=henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org,kjellander@google.com,ivoc@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1345983009 Cr-Commit-Position: refs/heads/master@{#9987}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.