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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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2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
mediasession.cc
45cc890
Assorted logging pedantry
by Jonas Olsson
· 7 years ago
8e545ee
Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."
by Tommi
· 7 years ago
6780c51
Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
by Joachim Bauch
· 7 years ago
e831b8c
Add MSID signaling compatibility for Unified Plan endpoints
by Steve Anton
· 7 years ago
3518e7b
Add the rejected TransportInfo when creating an answer.
by Zhi Huang
· 7 years ago
ad7bffc
Parameterize PeerConnection media tests for Unified Plan
by Steve Anton
· 7 years ago
fa2260d
Add support for data channels with Unified Plan
by Steve Anton
· 7 years ago
dcc3c02
Add support for JSEP offer/answer with transceivers
by Steve Anton
· 7 years ago
b1c1de1
Use the SDP ContentInfo helpers to avoid downcasting
by Steve Anton
· 7 years ago
5adfafd
Make ContentInfo/ContentDescription slightly more ergonomic
by Steve Anton
· 7 years ago
5634427
Remove unused properties from MediaContentDescription
by Steve Anton
· 7 years ago
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
7aee3d5
Fix ortc_api circular deps.
by Patrik Höglund
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
80cfb52
RTC_CHECK'ing content type before static_casting descriptions.
by Taylor Brandstetter
· 7 years ago
1c34974
Fixing invalid calls to FindMatchingCodec.
by Taylor Brandstetter
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 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.cc]
8ffb9c3
Change RtpSender to have multiple stream_ids
by Steve Anton
· 7 years ago
84f6a3f
Move optional.h to webrtc/api/
by kwiberg
· 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
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 7 years ago
8b7e9ad
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
by deadbeef
· 7 years ago
7914b8c
Negotiate the same SRTP crypto suites for every DTLS association formed.
by deadbeef
· 7 years ago
2f425aa
Fix SDP stream ID mismatch issue when a track's stream changes.
by deadbeef
· 7 years ago
eaa9c1d
Remove HAVE_SRTP define and unmaintained code.
by jbauch
· 7 years ago
e814a0d
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
by deadbeef
· 8 years ago
b789253
Accept SDP with TRANSPORT attributes missing from bundled m= sections.
by deadbeef
· 8 years ago
21e4e0b
Delete webrtc/base/common.h
by nisse
· 8 years ago
4b2e082
Use the same draft version in SDP data channel answers as used in the offer.
by zstein
· 8 years ago
c16fa5e
Replace all use of the VERIFY macro.
by nisse
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
ede5da4
Replace ASSERT by RTC_DCHECK in all non-test code.
by nisse
· 8 years ago
c80e741
Replace ASSERT(false) by RTC_NOTREACHED().
by nisse
· 8 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 8 years ago
352444f
RTC_[D]CHECK_op: Remove superfluous casts
by kwiberg
· 8 years ago
03d5fb1
Let MediaSession generate a FlexFEC SSRC when FlexFEC is active.
by brandtr
· 8 years ago
f823ede
Negotiate H264 profiles in SDP
by magjed
· 8 years ago
b05fa24
Optimize FindCodecById and ReferencedCodecsMatch
by magjed
· 8 years ago
3cf8ece
Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
by magjed
· 8 years ago
9f71ec5
Stop caching supported codecs in WebRtcVideoEngine2
by magjed
· 8 years ago
9fa4975
- Filter data channel codecs based on codec name instead of payload type, which may have been remapped.
by solenberg
· 8 years ago
4cedf2b
Add signaling to support ICE renomination.
by Honghai Zhang
· 8 years ago
1d7a637
Fixing off-by-one error with max SCTP id.
by Taylor Brandstetter
· 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
dc4eb8c
Refactoring some tests in peerconnectioninterface_unittest.cc.
by Taylor Brandstetter
· 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
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
5f0b83b
Enabling rtcp-rsize negotiation and fixing some issues with it.
by Taylor Brandstetter
· 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
· 9 years ago
0ed85b2
Track pending ICE restarts independently for different media sections.
by deadbeef
· 9 years ago
65c7f67
Fix license headers in webrtc/pc
by kjellander
· 9 years ago
9b8df25
Move talk/session/media -> webrtc/pc
by kjellander@webrtc.org
· 9 years ago
[Renamed (99%) from talk/session/media/mediasession.cc]
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
f475d36
Properly handle different transports having different SSL roles.
by Taylor Brandstetter
· 9 years ago
44f0819
Fixing bug where "mid" wasn't preserved across re-offers.
by deadbeef
· 9 years ago
1387149
Adding reduced size RTCP configuration down to the video stream level.
by deadbeef
· 9 years ago
b5cb19b
Fixing direction attribute in answer for non-RTP protocols.
by deadbeef
· 9 years ago
521ed7b
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 9 years ago
318166b
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 9 years ago
2764e10
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 9 years ago
c80741f
Fixing some issues with the direction attribute of m-lines in 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
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
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
· 9 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
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