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4 Commits

Author SHA1 Message Date
solenberg
d4cec0d8fa Remove MediaChannel::SetRemoteRenderer(). 
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1398823003

Cr-Commit-Position: refs/heads/master@{#10237}
 2015-10-09 15:55:54 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types. 
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
 2015-10-07 10:23:32 +00:00
deadbeef
70ab1a1ca8 Exposing RtpSenders and RtpReceivers from PeerConnection. 
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.

It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.

Review URL: https://codereview.webrtc.org/1351803002

Cr-Commit-Position: refs/heads/master@{#10100}
 2015-09-28 23:54:02 +00:00
deadbeef
6979b024d7 Adding stub files for RtpSender/RtpReceiver. 
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.

Review URL: https://codereview.webrtc.org/1364813004

Cr-Commit-Position: refs/heads/master@{#10065}
 2015-09-24 23:47:59 +00:00