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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2bb8447093a9c2899928960f623a361b30342fdb
/
pc
5f6bf24
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
by henrika
· 7 years ago
990d6b8
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
by Mirko Bonadei
· 7 years ago
90bace0
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
by henrika
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
b2d355e
Reland: Reject the description with fewer m= sections.
by Zhi Huang
· 7 years ago
074dece
Fix flaky DataChannel integration test
by Steve Anton
· 7 years ago
d5585ca
Move almost all references from WebRtcSession to PeerConnection
by Steve Anton
· 7 years ago
c4faa9c
Remove QUIC transport/data channel
by Steve Anton
· 7 years ago
ef48df9
Fix the issues in SrtpTransport.
by Zhi Huang
· 7 years ago
8a63f78
Rewrite the remaining few WebRtcSession tests.
by Steve Anton
· 7 years ago
da6c095
Rewrite WebRtcSession data channel tests as PeerConnection tests
by Steve Anton
· 7 years ago
6f25b09
Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests"
by Steve Anton
· 7 years ago
8d3444d
Reland "Rewrite WebRtcSession media tests as PeerConnection tests"
by Steve Anton
· 7 years ago
f2662f0
Revert "Rewrite WebRtcSession media tests as PeerConnection tests"
by Olga Sharonova
· 7 years ago
b49b661
Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests"
by Olga Sharonova
· 7 years ago
78609d5
Reland of BWE allocation strategy
by Alex Narest
· 7 years ago
6f72f56
Change return types of refcount methods.
by Niels Möller
· 7 years ago
096e367
Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
by Steve Anton
· 7 years ago
3df5dca
Rewrite WebRtcSession media tests as PeerConnection tests
by Steve Anton
· 7 years ago
3b80aac
Fix flaky memory leak in RemoteAudioSource
by Steve Anton
· 7 years ago
dc9ca93
Revert "BWE allocation strategy"
by Alex Narest
· 7 years ago
a5fbc23
BWE allocation strategy
by Alex Narest
· 7 years ago
39260c4
Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."
by Lu Liu
· 7 years ago
54d1da1
BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
by Alex Narest
· 7 years ago
1b0eae3
Don't call deprecated CreatePeerConnectionFactory() overloads
by Karl Wiberg
· 7 years ago
6592f2c
Removes more unused ADM APIs:
by henrika
· 7 years ago
8b35df7
Try re-enabling VoiceChannel::TestInit.
by Kári Tristan Helgason
· 7 years ago
ede9ca5
Rewrite WebRtcSession ICE integration tests as PeerConnection tests
by Steve Anton
· 7 years ago
d6b4819
PeerConnection::StartRtcEventLog: Improve callback memory safety
by Karl Wiberg
· 7 years ago
919dc2e
Removes fallback from Linux PulseAudio to ALSA.
by henrika
· 7 years ago
589ae45
Revert "Reject the subsequent offer with fewer m= sections."
by Tommi
· 7 years ago
a8264db
Reject the subsequent offer with fewer m= sections.
by Zhi Huang
· 7 years ago
f1c6db1
Rewrite WebRtcSession ICE tests as PeerConnection tests
by Steve Anton
· 7 years ago
99c3fe5
Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter
by Elad Alon
· 7 years ago
80cfb52
RTC_CHECK'ing content type before static_casting descriptions.
by Taylor Brandstetter
· 7 years ago
b140b9f
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
by Taylor Brandstetter
· 7 years ago
9e6565b
Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed
by Elad Alon
· 7 years ago
c5bb00b
PeerConnection end-to-end test with a non-builtin codec
by Karl Wiberg
· 7 years ago
bdcee28
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
by Jonas Oreland
· 7 years ago
933d8b0
Reland "Added PeerConnectionObserver::OnRemoveTrack."
by Henrik Boström
· 7 years ago
6c0c55c
Revert "Added PeerConnectionObserver::OnRemoveTrack."
by Alex Loiko
· 7 years ago
ba97ba7
Added PeerConnectionObserver::OnRemoveTrack.
by Henrik Boström
· 7 years ago
604427b
Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort"
by Guido Urdaneta
· 7 years ago
b23ed7f
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
by Jonas Oreland
· 7 years ago
6b63cd5
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
by Steve Anton
· 7 years ago
97a9f76
Add sdputils.h with useful functions for working with session descriptions
by Steve Anton
· 7 years ago
82eb3c4
Remove dead version of StartRtcEventLog
by Elad Alon
· 7 years ago
acb2417
Fix apparent copy/paste error in comment (PeerConnection)
by Elad Alon
· 7 years ago
84255bb
Add explicit includes of refcountedobject.h where it is used.
by Niels Möller
· 7 years ago
fb26f85
Revert "Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h.""
by Niels Moller
· 7 years ago
bf6937a
Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
by Niels Möller
· 7 years ago
e2d6a06
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
1af3d82
Revert "Reland "Clean up libjingle API dependencies.""
by Henrik Kjellander
· 7 years ago
9185aca
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
04eaa15
Change the flag when RtpTransport objects send packet.
by Zhi Huang
· 7 years ago
a32dd01
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
83ccca1
Create and use RtcEventLogOutput for output
by Elad Alon
· 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
8c0f7a7
Add GetRemoteAudioSSLCertificate() to PeerConnection
by Steve Anton
· 7 years ago
4a87e1c
Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead
by Elad Alon
· 7 years ago
d25fa78
Revert "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
by Niels Moller
· 7 years ago
b7239a9
Make rtc_base/refcount.h self contained, not including refcountedobject.h.
by Niels Möller
· 7 years ago
978b876
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
d4404c2
Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
34cdd2d
Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
by Fredrik Solenberg
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
581df61
Revert "Reland "Clean up libjingle API dependencies.""
by Patrik Höglund
· 7 years ago
5117b04
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
b526158
Move the TransportController from p2p/base to pc/.
by Zhi Huang
· 7 years ago
d8970db
Delete unneeded includes of fileutils.h
by Niels Möller
· 7 years ago
7bcfc3b
Revert "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
bf66794
Revert "Move clients of WebRtcSession to use PeerConnection"
by Alex Loiko
· 7 years ago
57fb315
Clean up libjingle API dependencies.
by Patrik Höglund
· 7 years ago
3dc4d4a
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
94286cb
Add base fixture and PeerConnection wrapper for unit tests
by Steve Anton
· 7 years ago
02e7a19
Remove unnecessary video factory references in PeerConnectionFactory
by Magnus Jedvert
· 7 years ago
cf990f5
Reland: Completed the functionalities of SrtpTransport.
by Zhi Huang
· 7 years ago
835cc0c
Remove unnecessary audio references in PeerConnectionFactory
by Magnus Jedvert
· 7 years ago
4e2deab
Only return stats for the most recent unsignaled audio stream.
by deadbeef
· 7 years ago
b19012e
Remove the support of fallback from DTLS to SDES.
by zhihuang
· 7 years ago
eb23e17
Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )
by zhihuang
· 7 years ago
1d4db39
Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ )
by henrika
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
d45aea8
Serialize "a=x-google-flag:conference".
by deadbeef
· 7 years ago
5ada7ac
If SRTP sessions exist, don't create new ones when applying answer.
by deadbeef
· 7 years ago
58b0316
Expose new video codec factories in the PeerConnectionFactory API
by Magnus Jedvert
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
563934e
Clean up dependencies of peerconnection_unittest.
by Patrik Höglund
· 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