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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
baaf911c80e15b51ed7cfa52b873ba8c0049a883
/
call
/
call_unittest.cc
3d2ed19
Remove Transport implementation from ChannelSend
by Fredrik Solenberg
· 6 years ago
179a392
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
by Piotr (Peter) Slatala
· 6 years ago
cc8e8bb
Pass the media transport from JsepTransportController to Call.
by Piotr (Peter) Slatala
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
ae4237e
Set ChannelReceive transport at construction time.
by Niels Möller
· 6 years ago
918f50c
Use absl::make_unique and absl::WrapUnique directly
by Karl Wiberg
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
e5447fb
Removed fake rtp transport controller send.
by Sebastian Jansson
· 7 years ago
fc8d26b
Reland "Moved BitrateConfig out of Call::Config."
by Sebastian Jansson
· 7 years ago
e4bf600
Revert "Moved BitrateConfig out of Call::Config."
by Lu Liu
· 7 years ago
5897fe2
Moved BitrateConfig out of Call::Config.
by Sebastian Jansson
· 7 years ago
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
fedc00c
Optional: Use nullopt and implicit construction in /call
by Oskar Sundbom
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
d319534
Move ADM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
f3850f6
Voice Engine: Require caller to supply an AudioDecoderFactory
by Karl Wiberg
· 7 years ago
a32dd01
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 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
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/call/call_unittest.cc]
5c8942a
Move PacedSender ownership to RtpTransportControllerSend.
by Stefan Holmer
· 7 years ago
db2a9fc
Wire up RTP keep-alive in ortc api.
by sprang
· 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
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
4b97980
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
by zstein
· 7 years ago
441718e
Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ )
by charujain
· 7 years ago
9641c13
Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
by zstein
· 7 years ago
c3d4b48
Store/restore RTP state for audio streams with same SSRC within a call
by ossu
· 7 years ago
8c96a14
Simple tests for Call::SetBitrateConfig.
by zstein
· 7 years ago
7cb69d5
This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).
by zstein
· 7 years ago
37e99fd
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
by kwiberg
· 7 years ago
1c07c70
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
by kwiberg
· 7 years ago
670a7f3
Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
by kwiberg
· 7 years ago
1724cfb
WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
by kwiberg
· 7 years ago
8313a6f
Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
by brandtr
· 8 years ago
1cfbd60
Generalize FlexfecReceiveStream::Config.
by brandtr
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
10111bc
Passed AudioMixer to AudioState::Config.
by aleloi
· 8 years ago
dd31071
Added an empty AudioTransportProxy to AudioState.
by aleloi
· 8 years ago
7602aab
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
by solenberg
· 8 years ago
25445d3
Integrate FlexfecReceiveStream with Call.
by brandtr
· 8 years ago
11a9cbf
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
by skvlad
· 8 years ago
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
29b1a8d
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
by ossu
· 8 years ago
b25345e
Replace scoped_ptr with unique_ptr in webrtc/call/
by kwiberg
· 9 years ago
3a94154
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
566ef24
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
by solenberg
· 9 years ago
0ccae13
Changed FakeVoiceEngine into a MockVoiceEngine.
by Fredrik Solenberg
· 9 years ago
4f4ec0a
Re-Land: Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
43e83d4
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
by solenberg
· 9 years ago
a457752
Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
c7a8b08
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
by solenberg
· 9 years ago