405 Commits

Author	SHA1	Message	Date
stefan	1d8a506405	Add a PacketOptions struct to webrtc::Transport. ... This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}	2015-10-02 10:39:40 +00:00
pbos	da903eaabb	Unify newapi::RtcpMode and RTCPMethod. ... BUG=webrtc:1695 R=solenberg@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1373903003 Cr-Commit-Position: refs/heads/master@{#10143}	2015-10-02 09:37:18 +00:00
pbos	2d566686a2	Unify Transport and newapi::Transport interfaces. ... BUG=webrtc:1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1369263002 Cr-Commit-Position: refs/heads/master@{#10096}	2015-09-28 16:59:36 +00:00
stefan	4fbd145dce	Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. ... In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest. BUG=webrtc:4836 Review URL: https://codereview.webrtc.org/1368943002 Cr-Commit-Position: refs/heads/master@{#10087}	2015-09-28 10:57:23 +00:00
Peter Boström	5c389d3e09	Split webrtc/video into webrtc/{audio,call,video}. ... Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}	2015-09-25 11:58:39 +00:00