Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2db1778d38064855f79ff76420de211e4c283540
/
pc
/
mediasession_unittest.cc
4e70a72
Replace MediaContentDirection with RtpTransceiverDirection
by Steve Anton
· 7 years ago
1d03a75
Remove cricket::RtpTransceiverDirection
by Steve Anton
· 7 years ago
6f36747
Use local codec parameters in the answer.
by Zhi Huang
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
98ea2da
Removing logging in unit test that was committed accidentally.
by Taylor Brandstetter
· 7 years ago
1c34974
Fixing invalid calls to FindMatchingCodec.
by Taylor Brandstetter
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/mediasession_unittest.cc]
8ffb9c3
Change RtpSender to have multiple stream_ids
by Steve Anton
· 7 years ago
1c378ed
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
by zhihuang
· 7 years ago
3c74766
Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ )
by olka
· 7 years ago
a77e6bb
Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
by zhihuang
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
5869f50
Support encrypted RTP extensions (RFC 6904)
by jbauch
· 7 years ago
8b7e9ad
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
by deadbeef
· 7 years ago
30952b4
Add "ice-option:trickle" to generated offers/answers.
by deadbeef
· 7 years ago
eaa9c1d
Remove HAVE_SRTP define and unmaintained code.
by jbauch
· 7 years ago
4b2e082
Use the same draft version in SDP data channel answers as used in the offer.
by zstein
· 7 years ago
abcef5d
Replace std::tr1::tuple by ::testing::tuple.
by ehmaldonado
· 7 years ago
c8ee882
Replace use of ASSERT in test code.
by nisse
· 8 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 8 years ago
03d5fb1
Let MediaSession generate a FlexFEC SSRC when FlexFEC is active.
by brandtr
· 8 years ago
b05fa24
Optimize FindCodecById and ReferencedCodecsMatch
by magjed
· 8 years ago
2675274
Remove cricket::VideoCodec with, height and framerate properties
by perkj
· 8 years ago
4cedf2b
Add signaling to support ICE renomination.
by Honghai Zhang
· 8 years ago
cb56065
Add support for GCM cipher suites from RFC 7714.
by jbauch
· 8 years ago
dedfd28
Support for two audio codec lists down into WebRtcVoiceEngine.
by ossu
· 8 years ago
075af92
Initial asymmetric codec support in MediaSessionDescription
by ossu
· 8 years ago
6f8d686
Remove use of RtpHeaderExtension and clean up
by isheriff
· 8 years ago
fd8be34
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 8 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
8f65cdf
Only generate one CNAME per PeerConnection.
by zhihuang
· 8 years ago
cf5b37c
Accept all the media profiles required by JSEP.
by zhihuang
· 8 years ago
8c011e5
Simple lint fixes
by terelius
· 8 years ago
555604a
Replace scoped_ptr with unique_ptr in webrtc/base/
by jbauch
· 8 years ago
d713e86
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
by zhihuang
· 8 years ago
67cf2c1
Removing `preference` field from `cricket::Codec`.
by deadbeef
· 8 years ago
b7f425a
Accept all the media profiles required by JSEP.
by zhihuang
· 8 years ago
3102294
Replace scoped_ptr with unique_ptr in webrtc/pc/
by kwiberg
· 8 years ago
6ec641b
Fixing some issues with payload type mappings.
by Taylor Brandstetter
· 8 years ago
f475277
Rename constants files in webrtc/{media,p2p}
by kjellander
· 8 years ago
65c7f67
Fix license headers in webrtc/pc
by kjellander
· 8 years ago
9b8df25
Move talk/session/media -> webrtc/pc
by kjellander@webrtc.org
· 8 years ago
[Renamed (99%) from talk/session/media/mediasession_unittest.cc]
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 8 years ago
0eb15ed
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
by kwiberg
· 9 years ago
44f0819
Fixing bug where "mid" wasn't preserved across re-offers.
by deadbeef
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
456696a
Reland Change WebRTC SslCipher to be exposed as number only
by Guo-wei Shieh
· 9 years ago
27dc29b
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
by guoweis
· 9 years ago
4fe3c9a
Change WebRTC SslCipher to be exposed as number only.
by guoweis
· 9 years ago
7cbd188
Remove GICE (again).
by Peter Thatcher
· 9 years ago
d12140a
Revert change which removes GICE.
by guoweis
· 9 years ago
3a14bf3
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers.
by Henrik Boström
· 9 years ago
2159b89
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
5bdafd4
Revert "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 minyuel
· 9 years ago
a5b273a
Fixing problems with RTP extension ID conflict resolution
by deadbeef
· 9 years ago
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
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
· 9 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
34f2a9e
Initialize SSL in unittest_main.cc.
by pbos@webrtc.org
· 10 years ago
a09a999
(Auto)update libjingle 73222930-> 73226398
by buildbot@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
79047f9
(Auto)update libjingle 62691533-> 62713454
by henrike@webrtc.org
· 10 years ago
b90991d
Update libjingle 62472237->62550414
by henrike@webrtc.org
· 10 years ago
704bf9e
(Auto)update libjingle 62063505-> 62278774
by henrike@webrtc.org
· 10 years ago
32f485b
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
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
0be6aa0
Update talk to 51314459
by sergeyu@chromium.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