Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
a9229043e357edfa882e80a226d4653079ee730d
/
pc
fcf79cc
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
by Åsa Persson
· 5 years ago
ead0ec9
Add firing of OnRemoveTrack and OnRenegotationNeeded during rollback
by Eldar Rello
· 5 years ago
86d053c
Use source_sets in component builds and static_library in release builds.
by Mirko Bonadei
· 5 years ago
8038541
Update the header extensions capabilities with mid, rid and rrid
by Florent Castelli
· 5 years ago
ac0a4cb
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Niels Möller
· 5 years ago
35214fc
Add missing RTC_EXPORT for the component build.
by Mirko Bonadei
· 5 years ago
ef0627f
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Mirko Bonadei
· 5 years ago
f8998cf
Add a turn port prune policy to keep the first ready turn port.
by Honghai Zhang
· 5 years ago
fbde32e
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
by Niels Möller
· 5 years ago
5ab79e6
Reland "Implement rollback for setRemoteDescription"
by Eldar Rello
· 5 years ago
5740f3e
Clarify expectation on GlobalLock
by Danil Chapovalov
· 5 years ago
d0704ce
Remove RTCP tests from channel_unittest.
by Bjorn A Mellem
· 5 years ago
907f154
Revert "Implement rollback for setRemoteDescription"
by Alex Loiko
· 5 years ago
16d4c4d
Implement rollback for setRemoteDescription
by Eldar Rello
· 5 years ago
f3f03e2
Removing outdated tests.
by Alex Loiko
· 5 years ago
ff27da5
Add/remove receive streams with SSRC 0 from media channels
by Saurav Das
· 5 years ago
bfcec4c
Delete old placeholders for moved api/ header files
by Niels Möller
· 5 years ago
8e1343a
Add an alt-protocol to SDP to indicate which m= sections use a plugin transport.
by Bjorn A Mellem
· 5 years ago
7da4e56
Allow receive-only use of datagram transport for data channels.
by Bjorn A Mellem
· 5 years ago
fc604aa
Unset sinks when deleting CompositeDataChannelTransport.
by Bjorn A Mellem
· 5 years ago
bc3eebc
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface.""
by Bjorn A Mellem
· 5 years ago
1b57541
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
by Mirko Bonadei
· 5 years ago
ef14f07
Delete AudioDecoder method IncomingPacket
by Niels Möller
· 5 years ago
988e63e
Proxy OnRtcpPacketReceived to the worker thread in channel tests.
by Bjorn A Mellem
· 5 years ago
d702231
Cleanup deprecated monitoring of MediaTransport state.
by Bjorn A Mellem
· 5 years ago
e942b14
New build target api:media_interface
by Niels Möller
· 5 years ago
1b83a9e
Only handle each RTCP once.
by Sebastian Jansson
· 5 years ago
7b04a91
Delete almost all default methods on PeerConnectionInterface
by Niels Möller
· 5 years ago
317a1f0
Use std::make_unique instead of absl::make_unique.
by Mirko Bonadei
· 5 years ago
7262fc2
Refactor Rtp Receivers to accept SSRC 0.
by Saurav Das
· 5 years ago
eaaaf41
Introduce api/crypto/BUILD.gn.
by Mirko Bonadei
· 5 years ago
e78fd80
New class DummyPeerConnection
by Niels Möller
· 5 years ago
01be33b
Using lambdas instead of rtc::Bind in BaseChannel.
by Sebastian Jansson
· 5 years ago
262bbae
Fix rare audioLevel flake in RTCStatsIntegrationTest.
by Henrik Boström
· 5 years ago
65f17ca
Move MediaTransportInterface out of the libjingle_peerconnection_api target
by Niels Möller
· 5 years ago
cc46b10
Add a usage pattern bit for host-host connections.
by Qingsi Wang
· 5 years ago
fcfeefe
Move rtc_error.{h,cc} to its own build target.
by Mirko Bonadei
· 5 years ago
437077d
Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface.""
by Qingsi Wang
· 5 years ago
487f9a1
Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
by Bjorn A Mellem
· 5 years ago
116ffe7
Switch to compiling WebRTC -std=c++14 by default
by Danil Chapovalov
· 5 years ago
cc62b16
Add qualityLimitationResolutionChanges stat
by Evan Shrubsole
· 5 years ago
be2e5f7
Make payload type demux conditional on media direction
by Steve Anton
· 5 years ago
20232a9
Use obfuscated IPs in logging in p2p/ and pc/.
by Qingsi Wang
· 5 years ago
662e31f
Prepare to move packet_socket_factory to api/.
by Patrik Höglund
· 5 years ago
340e0c5
Delete old version of PeerConnection::SetConfiguration
by Niels Möller
· 5 years ago
a837030
Split out RtpSource from libjingle_peerconnection_api
by Niels Möller
· 5 years ago
8b14b0d
Revert "Refactor SCTP data channels to use DataChannelTransportInterface."
by Henrik Boström
· 5 years ago
4c85828
Refactor SCTP data channels to use DataChannelTransportInterface.
by Bjorn A Mellem
· 5 years ago
25eb47c
Make the RtpHeaderParserImpl available to tests and tools only.
by Tommi
· 5 years ago
6dcd4dc
New target for api/rtp_parameters.h and api/media_types.h.
by Niels Möller
· 5 years ago
703ea95
Only create a datagram RTP transport if the datagram transport should be used for RTP.
by Bjorn A Mellem
· 5 years ago
fa046b3
Remove unused using statements in webrtc_sdp.cc
by Elad Alon
· 5 years ago
149dc72
Add support for RTCTransportStats.selectedCandidatePairChanges
by Jonas Oreland
· 5 years ago
0c141c5
Fix frames dropped statistics
by Johannes Kron
· 5 years ago
3c02842
Add TURN_LOGGING_ID
by Jonas Oreland
· 5 years ago
224c69d
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
by Niels Möller
· 5 years ago
b689af4
Changes to enable use of DatagramTransport as a data channel transport.
by Bjorn A Mellem
· 5 years ago
21e99da
Add implemented-but-missing members to RTCMediaStreamTrackStats::Members
by Henrik Boström
· 5 years ago
364b267
Replace DatagramDtlsAdaptor with DatagramRtpTransport.
by Bjorn A Mellem
· 5 years ago
2579f0c
RTCError as return type for PeerConnectionInterface::SetConfiguration
by Niels Möller
· 5 years ago
a0b52b5
Remove zhihuang@webrtc.org from OWNERS
by Steve Anton
· 5 years ago
1ba5dec
Reland "Set the usage pattern bits for adding remote ICE candidates from SDP."
by Qingsi Wang
· 5 years ago
6b43086
Reland "[GetStats] Expose video codec implementation in standardized metrics."
by Henrik Boström
· 5 years ago
fa8f4ee
Only combine media transport and ICE states if used for media.
by Bjorn A Mellem
· 5 years ago
df625f4
Revert "[GetStats] Expose video codec implementation in standardized metrics."
by Henrik Andreassson
· 5 years ago
2b9fa09
[GetStats] Expose video codec implementation in standardized metrics.
by Henrik Boström
· 5 years ago
bbeb109
Reporting audio device underrun counter
by Alex Narest
· 5 years ago
d419808
Revert "Set the usage pattern bits for adding remote ICE candidates from SDP."
by Qingsi Wang
· 5 years ago
7c6f74a
Set the usage pattern bits for adding remote ICE candidates from SDP.
by Qingsi Wang
· 5 years ago
9cfdb20
Control PeerConnectionFactory's default min/starting/max bitrates from experiment
by Elad Alon
· 5 years ago
43faee0
Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing
by Alex Drake
· 5 years ago
78a7138
Remove MediaTransport from Call.
by Tommi
· 5 years ago
5b5d97c
Reland of "Reporting of decoding_codec_plc events""
by Alex Narest
· 5 years ago
83bbe91
Delete deprecated rtc_event_log header
by Danil Chapovalov
· 5 years ago
00c7ecf
Surface CandidatePairChange event
by Alex Drake
· 5 years ago
6563934
Revert "Sanitize the codec list before sending it to the media engine"
by Artem Titov
· 5 years ago
032ea9c
Remove "Missing track ID" StatsCollector log message
by Steve Anton
· 5 years ago
add7ef9
Sanitize the codec list before sending it to the media engine
by Steve Anton
· 5 years ago
928e7a3
Make ID of datachannel stats not depend on dc.id
by Harald Alvestrand
· 5 years ago
bedb7a8
Revert "Reporting of decoding_codec_plc events"
by Mirko Bonadei
· 5 years ago
0a88ea0
Reporting of decoding_codec_plc events
by Alex Narest
· 5 years ago
cb11a31
Guard GenerateUniqueId() against concurrent access.
by Yves Gerey
· 5 years ago
1a13c8f
Add option to remove transport sequence number from FEC packet calculation
by Anton Sukhanov
· 5 years ago
97321b6
Adds test for experimental remote estimate SDP negotiation.
by Sebastian Jansson
· 5 years ago
e1795f4
Adds remote estimate RTCP packet.
by Sebastian Jansson
· 5 years ago
f781bb5
[Unit test] Add check to prevent segfault on empty vector.
by Yves Gerey
· 5 years ago
2ab97f6
Migrate WebRTC test infra to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
79b6980
[PeerConnection] Implement restartIce().
by Henrik Boström
· 5 years ago
0182a03
Reland "Remove the injectable bitrate allocation strategy API."
by Jonas Olsson
· 5 years ago
5a29d52
Propagate datagram SentNotification for RTCP packets
by Anton Sukhanov
· 5 years ago
9a44b2d
Add an end-to-end integration test for |enable_encrypted_rtp_header_extensions|
by Steve Anton
· 5 years ago
e95b57c
Revert "Remove the injectable bitrate allocation strategy API."
by Mirko Bonadei
· 5 years ago
3ae59d3
Use the dummy address 0.0.0.0:9 in the c= and the m= lines if the
by Qingsi Wang
· 5 years ago
ac6c096
Integrate datagram feedback loop
by Anton Sukhanov
· 5 years ago
ee303fa
Move datagram_dtls_adaptor from p2p/base/ to pc/
by Anton Sukhanov
· 5 years ago
80cb3f6
Remove the injectable bitrate allocation strategy API.
by Jonas Olsson
· 5 years ago
a4d8737
Format almost everything.
by Jonas Olsson
· 5 years ago
3fbf1e2
Reduce kMaxSimulcastStreams to 3
by Florent Castelli
· 5 years ago
d2c336f
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
by Henrik Boström
· 5 years ago
53d45ba
Make TaskQueueFactory required construction parameter for Call
by Danil Chapovalov
· 5 years ago
Next »