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gerrit-public.fairphone.software
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platform
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external
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webrtc
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93a7b2470fe7ef74abde16ed84b89bb88418d9df
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pc
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mediasession.cc
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081f34b
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
by Peter Thatcher
· 9 years ago
fa30180
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
by pthatcher
· 9 years ago
3449faa
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
by Peter Thatcher
· 9 years ago
083b73f
Use std::string references instead of copying contents.
by jbauch
· 9 years ago
f393829
Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.
by deadbeef
· 9 years ago
2e7a098
Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
by Noah Richards
· 9 years ago
2d25b44
Check associated payload type when negotiate RTX codecs.
by changbin.shao@webrtc.org
· 10 years ago
a747093
After another round of reviews.
by lally@webrtc.org
· 10 years ago
ec97c65
Attempt on read-only acceptance of -12.
by lally@webrtc.org
· 10 years ago
586f2ed
Change GetStreamBySsrc to not copy StreamParams.
by tommi@webrtc.org
· 10 years ago
5ad4178
Move the Jingle-specific network code into webrtc/libjingle.
by pthatcher@webrtc.org
· 10 years ago
269fb4b
move xmpp and p2p to webrtc
by henrike@webrtc.org
· 10 years ago
f15dee6
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
by tommi@webrtc.org
· 10 years ago
28100cb
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
by henrike@webrtc.org
· 10 years ago
d1ba6d9
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
by henrike@webrtc.org
· 10 years ago
742922b
Make the media content send only if offerToReceive is false while local streams exist.
by jiayl@webrtc.org
· 10 years ago
7d4891d
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
by jiayl@webrtc.org
· 10 years ago
c172320
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
by jiayl@webrtc.org
· 10 years ago
52055a2
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
by jiayl@webrtc.org
· 10 years ago
a09a999
(Auto)update libjingle 73222930-> 73226398
by buildbot@webrtc.org
· 10 years ago
56d8e05
A followup to r6828 to fix a condition check in mediasession.cc.
by jiayl@webrtc.org
· 10 years ago
e7d47a1
Maintain the order of the m-lines in CreateOffer and CreateAnswer.
by jiayl@webrtc.org
· 10 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 10 years ago
ff1b1bf
When creating an answer, takes the codec preference from the offer.
by wu@webrtc.org
· 10 years ago
8dcd43c
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
by jiayl@webrtc.org
· 10 years ago
9c16c39
Sets the SCTP port codec in the native SessionDescription.
by jiayl@webrtc.org
· 11 years ago
79047f9
(Auto)update libjingle 62691533-> 62713454
by henrike@webrtc.org
· 11 years ago
b90991d
Update libjingle 62472237->62550414
by henrike@webrtc.org
· 11 years ago
4b26e2e
Update libjingle to 59676287
by sergeyu@chromium.org
· 11 years ago
cecfd18
Update talk to 55821645.
by wu@webrtc.org
· 11 years ago
97077a3
Update libjingle to 55618622. Update libyuv to r826.
by wu@webrtc.org
· 11 years ago
19f27e6
Update talk to 54527154.
by mallinath@webrtc.org
· 11 years ago
7818752
Update libjingle to 53856368.
by wu@webrtc.org
· 11 years ago
a27be8e
Update libjingle to CL 53398036.
by mallinath@webrtc.org
· 11 years ago
1112c30
Update libjingle to 53057474.
by mallinath@webrtc.org
· 11 years ago
1e09a71
Update talk folder to revision=49952949
by henrike@webrtc.org
· 11 years ago
28654cb
Update talk folder to revision=49713299.
by henrike@webrtc.org
· 11 years ago
28e2075
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk
by henrike@webrtc.org
· 11 years ago