webrtc_m130

History

mflodman dc7d0d2ef0 Move, almost, all receive side references to RTP to RtpStreamReceiver.

There are still a few places in VideoReceiveStream where the RTP module
is explicitly used, e.g. setting up a/v sync, but it's a bigger task to
change and that will be done in a follow up instead of in this CL.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/1947913002
Cr-Commit-Position: refs/heads/master@{#12642}

2016-05-06 12:32:30 +00:00

include

Move, almost, all receive side references to RTP to RtpStreamReceiver.

2016-05-06 12:32:30 +00:00

mocks

Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream.

2016-04-15 12:59:21 +00:00

source

Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.

2016-05-03 06:44:11 +00:00

test

Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/