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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
988cc0870bb49c855b4547f929462eeb38186d12
/
call
/
rampup_tests.h
f5e767d
Don't send max allocation probe unless allocation changed.
by Sebastian Jansson
· 6 years ago
75e3647
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
by Artem Titov
· 6 years ago
631cafa
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
by Artem Titov
· 6 years ago
46c4e60
Introduce SimulatedNetworkReceiverInterface.
by Artem Titov
· 6 years ago
7258224
Replaces call config create in tests with modify.
by Sebastian Jansson
· 6 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/rampup_tests.h]
413ee9a
Use SingleThreadedTaskQueue in DirectTransport
by eladalon
· 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
d8ce1e1
Move SelectMediaType from RampUpTester to BaseTest.
by nisse
· 7 years ago
e5ad5ca
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
by nisse
· 7 years ago
45b5fe5
Don't report perf metrics for packet loss ramp-up tests.
by stefan
· 8 years ago
0f8b403
Introduce a new constructor to PlatformThread.
by tommi
· 8 years ago
5ef2bc1
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
by philipel
· 8 years ago
b80bdca
Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ )
by philipel
· 8 years ago
a518a39
Fixes a bug where a video stream can get stuck in the suspended state.
by stefan
· 8 years ago
5a2c506
Set the start bitrate to the delay-based BWE.
by stefan
· 8 years ago
38d8b3c
Clean up ramp-up tests and make sure they all pass.
by stefan
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago
db752f9
Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )"
by stefan
· 8 years ago
fbfb536
Explicitly enable RED over RTX in rampup tests.
by brandtr
· 8 years ago
11a9cbf
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
by skvlad
· 8 years ago
fa10b55
Releand of Let ViEEncoder handle resolution changes.
by perkj
· 8 years ago
3b703ed
Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
by perkj
· 8 years ago
26105b4
Let ViEEncoder handle resolution changes.
by perkj
· 8 years ago
86cc6ff
Variable audio bitrate.
by mflodman
· 8 years ago
b25345e
Replace scoped_ptr with unique_ptr in webrtc/call/
by kwiberg
· 8 years ago
ff2a635
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 9 years ago
d20e651
Fix test bug introduced in r11101.
by Stefan Holmer
· 9 years ago
e74eef1
Add CreateSend/ReceiveTransport() methods to CallTest.
by stefan
· 9 years ago
ff48361
Step 1 to prepare call_test.* for combined audio/video tests.
by stefan
· 9 years ago
[Renamed (82%) from webrtc/video/rampup_tests.h]
5811a39
Replace EventWrapper in video/, test/ and call/.
by Peter Boström
· 9 years ago
8c38e8b
Clean up PlatformThread.
by Peter Boström
· 9 years ago
12411ef
Move ThreadWrapper to ProcessThread in base.
by pbos
· 9 years ago
98f5351
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
f116bd0
Call OnSentPacket for all packets sent in the test framework.
by stefan
· 9 years ago
092508a
Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc.
by stefan
· 9 years ago
4fbd145
Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
by stefan
· 9 years ago
5c389d3
Split webrtc/video into webrtc/{audio,call,video}.
by Peter Boström
· 9 years ago
6b8d355
Reland "Wire up send-side bandwidth estimation."
by Erik Språng
· 9 years ago
c9bbeb0
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
by Erik Språng
· 9 years ago
ef165ee
Wire up send-side bandwidth estimation.
by sprang
· 9 years ago
68786d2
Wire up PacketTime to ReceiveStreams.
by stefan
· 9 years ago
11324b9
Wait for a longer time (5 seconds) before establishing the first bandwidth estimate.
by Stefan Holmer
· 9 years ago
468e62a
Remove MimdRateControl and factories for RemoteBitrateEstimor.
by Erik Språng
· 9 years ago
f2f8283
Use rtc::CriticalSection in webrtc/video/.
by Peter Boström
· 9 years ago
23fba1f
Add AudioReceiveStream to Call API.
by Fredrik Solenberg
· 9 years ago
e62202f
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
by Shao Changbin
· 9 years ago
14665ff
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
by kjellander@webrtc.org
· 10 years ago
00b8f6b
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
by kwiberg@webrtc.org
· 10 years ago
8f27fcc
Revert 8028 "Support associated payload type when registering Rt..."
by andrew@webrtc.org
· 10 years ago
2a16964
Support associated payload type when registering Rtx payload type.
by pbos@webrtc.org
· 10 years ago
273a414
Report encoded frame size in VideoSendStream.
by pbos@webrtc.org
· 10 years ago
3d7da88
Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.
by stefan@webrtc.org
· 10 years ago