| d3c944755ec546f46d5bdd84bff359fe6c4639e9 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Nuke TickTime::UseFakeClock.
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.
BUG=webrtc:5318
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512853002 .
Cr-Commit-Position: refs/heads/master@{#10947}
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
|
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 23fba1ffa0079f70744a83bcf4e85501dc226013 |
|
29-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51749004
Cr-Commit-Position: refs/heads/master@{#9114}
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 1ccff349ee2c7023a1124346ba068689238cc33d |
|
06-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix crashing fake network pipe tests.
These tests are not included in bots, this will be fixed in a follow-up by pbos@webrtc.org.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6837 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| b8e9e44eac26ab50c07161c55643e1e442927709 |
|
09-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add full stack test cases with a fake network pipe.
R=pbos@webrtc.org
BUG=1872
Review URL: https://webrtc-codereview.appspot.com/20889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| caba2d2a370cb6b5e67c881ecfa57fdac7411de8 |
|
14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add DeliveryStatus enum to DeliverPacket().
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| 8f69330310bf786cff373c225967e7459fb0b560 |
|
26-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace scoped_array<T> with scoped_ptr<T[]>.
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| c0e9aebe8f11e8622dc146406d8263f4bb436008 |
|
26-Feb-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SetConfig method to FakeNetworkPipe and to DirectTransport
This method allow the user to change the network configuration
during run-time. This is useful when testing how components react
to changing bandwidth.
BUG=2636
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|
| faada6e604e04e0765f93829cd29782667a1f235 |
|
18-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Integrate fake_network_pipe into direct_transport.
TEST=trybots
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/fake_network_pipe_unittest.cc
|