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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
channelmanager.cc
2dfc42d
Prepare to make BaseChannel depend on RtpTransportInternal only.
by Zhi Huang
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
8699a32
Have BaseChannel take MediaChannel as unique_ptr
by Steve Anton
· 7 years ago
c9e1560
Modernize and cleanup ChannelManager
by Steve Anton
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
b140b9f
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
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/channelmanager.cc]
774115c
Change ChannelManager to use unique_ptr
by Steve Anton
· 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
2f45b6b
Remove unused "crypto_options_" field.
by jbauch
· 7 years ago
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 7 years ago
eaabdf6
Delete MediaController class, move Call ownership to PeerConnection.
by nisse
· 7 years ago
7914b8c
Negotiate the same SRTP crypto suites for every DTLS association formed.
by deadbeef
· 7 years ago
d48f488
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
by jbauch
· 8 years ago
e814a0d
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
by deadbeef
· 8 years ago
1a2183d
Removing unnecessary parameters from CreateXChannel methods.
by deadbeef
· 8 years ago
112b2e9
Switching some interfaces to use std::unique_ptr<>.
by deadbeef
· 8 years ago
7d25426
Delete unneeded includes of base/common.h.
by nisse
· 8 years ago
f534659
Adding ability for BaseChannel to use PacketTransportInterface.
by deadbeef
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
b2cdd93
Remove the dependency of TransportChannel and TransportChannelImpl.
by zhihuang
· 8 years ago
6ce9259
Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ )
by zhihuang
· 8 years ago
5aed06c
make the DtlsTransportWrapper inherit form DtlsTransportInternal
by zhihuang
· 8 years ago
ac22f70
Refactoring of RTCP options in BaseChannel.
by deadbeef
· 8 years ago
f5b251b
Remove BaseChannel's dependency on TransportController.
by zhihuang
· 8 years ago
ede5da4
Replace ASSERT by RTC_DCHECK in all non-test code.
by nisse
· 8 years ago
953c2ce
Reland of: Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
c0dad89
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
by deadbeef
· 8 years ago
67b3bbe
Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 8 years ago
ebbe4f2
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
by zhihuang
· 8 years ago
ffc6118
Don't cache video codec list in VideoEngine2.
by brandtr
· 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
cb56065
Add support for GCM cipher suites from RFC 7714.
by jbauch
· 8 years ago
14d5dbe
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
by ivoc
· 8 years ago
9e03c3b
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
by ivoc
· 8 years ago
1895526
Move RtcEventLog object from inside VoiceEngine to Call.
by Ivo Creusen
· 8 years ago
05b9803
Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface.
by solenberg
· 8 years ago
dedfd28
Support for two audio codec lists down into WebRtcVoiceEngine.
by ossu
· 8 years ago
5d97a9a
Adding more detail to MessageQueue::Dispatch logging.
by Taylor Brandstetter
· 8 years ago
6c87a67
Do not create a temporary transport channel when using max-bundle
by skvlad
· 8 years ago
c1513ee
Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API.
by ivoc
· 8 years ago
33b01f2
Adds network thread to rtc::BaseChannel
by Danil Chapovalov
· 8 years ago
ff97631
- Add temporary VoEBase::audio_device_module() method.
by solenberg
· 8 years ago
b252856
Remove all uses of the HAVE_CONFIG_H define.
by Henrik Kjellander
· 8 years ago
c11b184
Remove CaptureManager and related calls in ChannelManager.
by perkj
· 8 years ago
fb45d17
Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/)
by Per
· 9 years ago
74622e0
Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ )
by perkj
· 9 years ago
e9c0cdf
Removed unused cricket::VideoCapturer methods:
by perkj
· 9 years ago
a509241
This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting.
by Per
· 9 years ago
51542be
Introduce struct MediaConfig, with construction-time settings.
by nisse
· 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/channelmanager.cc]
162c339
Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #14 id:300001 of https://codereview.webrtc.org/1655793003/ )
by perkj
· 9 years ago
4d19c5b
This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it.
by Per
· 9 years ago
4b2a5a8
Revert of Make cricket::VideoCapturer implement VideoSourceInterface (patchset #12 id:260001 of https://codereview.webrtc.org/1655793003/ )
by perkj
· 9 years ago
2f21789
This cl introduce a VideoSourceInterface and let cricket::VideoCapturer implement it.
by perkj
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
e73afba
New rtc::VideoSinkInterface.
by nisse
· 9 years ago
2098fca
Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ )
by nisse
· 9 years ago
a862d45
New rtc::VideoSinkInterface.
by Niels Möller
· 9 years ago
d66b44d
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 9 years ago
a4df27b
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
by ivoc
· 9 years ago
f4f5cb0
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 9 years ago
36d4c54
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
by ivoc
· 9 years ago
ae2c5ad
Added option to specify a maximum file size when recording an AEC dump.
by ivoc
· 9 years ago
822bdf9
Remove cricket::VideoEncoderConfig.
by Peter Boström
· 9 years ago
1a9d615
Add tracing to public PeerConnection methods.
by Peter Boström
· 9 years ago
246b817
Refactor handling of AudioOptions.
by solenberg
· 9 years ago
9d69c3f
Return a copy of the supported RTP header extensions instead of a reference.
by Stefan Holmer
· 9 years ago
bd13838
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
by solenberg
· 9 years ago
797ef12
Added StopAecDump function to PeerConnectionFactory.
by ivoc
· 9 years ago
112a3d8
Added functions on libjingle API to start and stop the recording of an RtcEventLog.
by ivoc
· 9 years ago
dfc8f4f
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
by solenberg
· 9 years ago
4a3ccad
Remove SetAudioDelayOffset() and friends.
by solenberg
· 9 years ago
61e933e
Remove ChannelManager::GetCapabilities()
by solenberg
· 9 years ago
facbbec
Remove use of DeviceManager from ChannelManager.
by solenberg
· 9 years ago
cbecd35
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
by deadbeef
· 9 years ago
7d17336
Remove the [Un]RegisterVoiceProcessor() API.
by Fredrik Solenberg
· 9 years ago
a81a42f
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
by torbjorng
· 9 years ago
47ee2f3
TransportController refactoring.
by deadbeef
· 9 years ago
c1a1b35
Remove the SetLocalMonitor() API.
by solenberg
· 9 years ago
8902433
Revert "TransportController refactoring."
by Guo-wei Shieh
· 9 years ago
9af63f4
TransportController refactoring.
by deadbeef
· 9 years ago
709ed67
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
by Fredrik Solenberg
· 9 years ago
c232096
Remove cricket::VideoProcessor and AddVideoProcessor() functionality
by Magnus Jedvert
· 9 years ago
0c02264
Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.
by Fredrik Solenberg
· 9 years ago
c28a896
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
by Jelena Marusic
· 9 years ago
ccb49e7
Remove Soundclip handling from libjingle.
by Fredrik Solenberg
· 9 years ago
4b60c73
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
by Fredrik Solenberg
· 9 years ago
81ea54e
Remove WebRtcVideoEngine.
by Peter Boström
· 9 years ago
7fb711f
Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class.
by Fredrik Solenberg
· 9 years ago
56d5028
Remove SignalCaptureStateChange from MediaEngine.
by Peter Thatcher
· 9 years ago
77f0e3f
Remove GetStartCaptureFormat and some related code.
by Peter Thatcher
· 9 years ago
4aef5fe
Add thread checks to the CaptureManager.
by hbos@webrtc.org
· 10 years ago
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