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gerrit-public.fairphone.software
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platform
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external
/
webrtc
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f3850f6933c468625511fc9e2df9aa4b9fefed89
/
call
/
BUILD.gn
f3850f6
Voice Engine: Require caller to supply an AudioDecoderFactory
by Karl Wiberg
· 7 years ago
245660a
Fix Gn untracked headers in webrtc/call.
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/call/BUILD.gn]
84f6a3f
Move optional.h to webrtc/api/
by kwiberg
· 7 years ago
529662a
Move array_view.h to webrtc/api/
by kwiberg
· 7 years ago
334f9e6
Tracking mock_paced_sender.h with a GN target
by mbonadei
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
95c8f65
Now that https://codereview.webrtc.org/3003643002 is landed we can
by mbonadei
· 7 years ago
440b6d9
Move video send/receive stream headers to webrtc/call.
by aleloi
· 7 years ago
f3f5c0e
Change ThreadChecker to SequencedTaskChecker in internal::Call
by eladalon
· 7 years ago
b332917
Rename RsidResolutionObserver to SsrcBindingObserver.
by Steve Anton
· 7 years ago
db2a9fc
Wire up RTP keep-alive in ortc api.
by sprang
· 7 years ago
5166e54
Tracking mock_process_thread with a GN target
by mbonadei
· 7 years ago
e2173d9
Only one implementation of MockRtpPacketSink once
by eladalon
· 7 years ago
f6a861a
Remove remains of webrtc/base
by ehmaldonado
· 7 years ago
c024740
Use relative paths in GN files.
by jianjun.zhu
· 7 years ago
370dd47
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
by ehmaldonado
· 7 years ago
9483b49
Remove remains of webrtc/base
by ehmaldonado
· 7 years ago
a52722f
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
by eladalon
· 7 years ago
0e7e786
Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
by guidou
· 7 years ago
cb83bdf
Create RtcpDemuxer. Capabilities:
by eladalon
· 7 years ago
0f15f92
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
by nisse
· 7 years ago
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 7 years ago
d76b7b2
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
by nisse
· 7 years ago
760a076
Create unit tests for RtpDemuxer
by eladalon
· 7 years ago
c3d4b48
Store/restore RTP state for audio streams with same SSRC within a call
by ossu
· 7 years ago
eed52bf
New class RtxReceiveStream.
by nisse
· 7 years ago
e4bcd6d
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
by nisse
· 7 years ago
2d9d21f
Add untracked headers in modules/rtp_rtcp
by danilchap
· 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
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 7 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 7 years ago
81c79f5
Creating webrtc:video_stream_api
by mbonadei
· 7 years ago
e0629c0
GN: Tighten up test target visibility + refactorings
by kjellander
· 7 years ago
cae45d0
Move RtpTransportControllerSend to a new file.
by nisse
· 7 years ago
8d609f6
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 7 years ago
37e99fd
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
by kwiberg
· 7 years ago
fbcc5cb
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 7 years ago
292084c
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 7 years ago
c5d62e2
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
by sprang
· 7 years ago
b8f9a32
Define RtpTransportControllerSendInterface.
by nisse
· 7 years ago
16ccfdf
Reland Move fake_audio_device to its own target.
by perkj
· 8 years ago
2f1a555
Enable GN check for webrtc/call
by kjellander
· 8 years ago
38cc1d6
Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
by nisse
· 8 years ago
3ebbcb5
Stop using VoEVideoSync in Call/VideoReceiveStream.
by solenberg
· 8 years ago
9aa3f0a
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
by mbonadei
· 8 years ago
69dc7db
Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
by mbonadei
· 8 years ago
35a3270
Moving webrtc.gni up one level from build/
by mbonadei
· 8 years ago
021eef3
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
by ehmaldonado
· 8 years ago
5fbcd22
Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ )
by danilchap
· 8 years ago
0b5a26a
Refactor webrtc_perf_tests into several source_sets.
by ehmaldonado
· 8 years ago
7250b39
Move FlexfecReceiveStream from api/call/ to call/.
by brandtr
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
a8eb756
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 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
bf6a45b
Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency
by charujain
· 8 years ago
76648da
Add FlexfecReceiveStream.
by brandtr
· 8 years ago
e40a7ee
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
by kjellander
· 8 years ago
cc91d28
Moved RtcEventLog files from call/ to logging/
by skvlad
· 8 years ago
89a3a1a
Moved Gn target rtc_event_log to one directory above.
by charujain
· 8 years ago
b62dbbe
GN: Change rtc_source_set targets --> rtc_static_library
by kjellander
· 8 years ago
e9cac75
Reenabled the RtcEventLog unittests
by skvlad
· 8 years ago
e9cc686
GN Templates: Move common_inherited_config to the template.
by ehmaldonado
· 8 years ago
7a2ce0b
GN Templates: Move common_config to the template.
by ehmaldonado
· 8 years ago
38a2132
GN: Introduce templates.
by ehmaldonado
· 8 years ago
26091b1
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
by perkj
· 8 years ago
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
8eb37a3
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
by perkj
· 8 years ago
cc16836
- Add task queue to Call with the intent of replacing the use of one of the process threads.
by perkj
· 8 years ago
0208322
GN: Add video_engine_tests
by Peter Boström
· 8 years ago
14897d0
Add missing dependencies on audio, video and call to the new GN files.
by katrielc
· 8 years ago
80e1207
Move congestion controller to a separate module.
by Stefan Holmer
· 9 years ago
0e7e259
Move BitrateAllocator from BitrateController logic to Call.
by mflodman
· 9 years ago
0c478b3
Rename ChannelGroup to CongestionController and move to webrtc/call/.
by mflodman
· 9 years ago
5c389d3
Split webrtc/video into webrtc/{audio,call,video}.
by Peter Boström
· 9 years ago