ff2a6351e0ad81ef8123c368fc17eeab40e66c71 |
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14-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Add ramp-up tests for transport sequence number with and w/o audio. Also add a perf metric tracking the average network latency. The audio stream test is disabled for now since audio isn't included in bitrate allocation. BUG=webrtc:5263 R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1582833002 . Cr-Commit-Position: refs/heads/master@{#11244}
/external/webrtc/webrtc/test/direct_transport.cc
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5811a39f14fd77ebc0793ee93d03ee15a669bd8f |
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10-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Replace EventWrapper in video/, test/ and call/. Makes use of rtc::Event which is simpler and can be used without allocating additional objects on the heap. Does not modify test/channel_transport/. BUG= R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1487893004 . Cr-Commit-Position: refs/heads/master@{#10968}
/external/webrtc/webrtc/test/direct_transport.cc
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d3c944755ec546f46d5bdd84bff359fe6c4639e9 |
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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/direct_transport.cc
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8c38e8b9b96d72317d6ce94c1442113b4e385dcb |
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26-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Clean up PlatformThread. * Move PlatformThread to rtc::. * Remove ::CreateThread factory method. * Make non-scoped_ptr from a lot of invocations. * Make Start/Stop void. * Remove rtc::Thread priorities, which were unused and would collide. * Add ::IsRunning() to PlatformThread. BUG= R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1476453002 . Cr-Commit-Position: refs/heads/master@{#10812}
/external/webrtc/webrtc/test/direct_transport.cc
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12411ef40e08c5e28ccde54ab3418c96676ffcbc |
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23-Nov-2015 |
pbos <pbos@webrtc.org> |
Move ThreadWrapper to ProcessThread in base. Also removes all virtual methods. Permits using a thread from rtc_base_approved (namely event tracing). BUG=webrtc:5158 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1469013002 Cr-Commit-Position: refs/heads/master@{#10760}
/external/webrtc/webrtc/test/direct_transport.cc
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98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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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/direct_transport.cc
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f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
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27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework. Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1419193002 Cr-Commit-Position: refs/heads/master@{#10430}
/external/webrtc/webrtc/test/direct_transport.cc
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1d8a506405734d0cef9653704b036ca4f1388960 |
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02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport. This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/webrtc/test/direct_transport.cc
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f2f828374c3ee1e1834c72bb27eaae88ef67bb40 |
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01-May-2015 |
Peter Boström <pbos@webrtc.org> |
Use rtc::CriticalSection in webrtc/video/. Removes heap allocation from CriticalSection creation. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50839004 Cr-Commit-Position: refs/heads/master@{#9126}
/external/webrtc/webrtc/test/direct_transport.cc
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38492c5b6fbb615159fa32b9cc24cd887295573b |
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22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Re-land 8810 "- Add a SetPriority method to ThreadWr..." > Revert 8810 "- Add a SetPriority method to ThreadWrapper" > Seeing if this is causing roll issues. > > > - Add a SetPriority method to ThreadWrapper > > - Remove 'priority' from CreateThread and related member variables from implementations > > - Make supplying a name for threads, non-optional > > > > BUG= > > R=magjed@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/44729004 > > TBR=tommi@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/48609004 TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50459005 Cr-Commit-Position: refs/heads/master@{#8819} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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90a1cb463092c5189b1a69837731a3395d79f61c |
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22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8810 "- Add a SetPriority method to ThreadWrapper" Seeing if this is causing roll issues. > - Add a SetPriority method to ThreadWrapper > - Remove 'priority' from CreateThread and related member variables from implementations > - Make supplying a name for threads, non-optional > > BUG= > R=magjed@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/44729004 TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48609004 Cr-Commit-Position: refs/heads/master@{#8818} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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b6817d793fa647ec77aaaaf74df82a94e46632bb |
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20-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
- Add a SetPriority method to ThreadWrapper - Remove 'priority' from CreateThread and related member variables from implementations - Make supplying a name for threads, non-optional BUG= R=magjed@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44729004 Cr-Commit-Position: refs/heads/master@{#8810} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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86639737b83d8877abc4810100e30a8af863189d |
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13-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove thread id from ThreadWrapper::Start(). Removes ThreadPosix::InitParams and a corresponding wait for an event. This unblocks ThreadPosix::Start which had to wait for thread scheduling for an event to trigger on the spawned thread, giving faster Start() calls. BUG=4413 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43699004 Cr-Commit-Position: refs/heads/master@{#8709} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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d5ce2e63dfef7d3744f16ef23174aef23dd72e8f |
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13-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove EventWrapper::Reset(). This simplifies the event wrapper which we've recently found issues in. Also refactoring EndToEndTest.RespectsNetworkState to not depend on it. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41939004 Cr-Commit-Position: refs/heads/master@{#8366} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
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15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. This fixes a variety of MSVC warnings about value truncations when implicitly storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and removes the need for a number of explicit casts. This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack". BUG=chromium:81439 TEST=none R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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c0e9aebe8f11e8622dc146406d8263f4bb436008 |
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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/direct_transport.cc
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faada6e604e04e0765f93829cd29782667a1f235 |
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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/direct_transport.cc
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27326b6a42e8dc2431390aa016cbebfa2151aa05 |
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20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename newapi::Transport::SendRTP()->SendRtp(). Also fit rampup_tests.cc to use internal::TransportAdapter instead of implementing its own. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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b082ade3db118113bba284d0f8fd32901371a2a0 |
|
18-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Hook up audio/video sync to Call. Adds an end-to-end audio/video sync test. BUG=2530, 2608 TEST=trybots R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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69969e2e2f0420df2765ab72d8e6f96d6d9d5d9c |
|
15-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improve Call tests for RTX. Also does some refactoring to reuse RtpRtcpObserver. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
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16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
|
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Separate Call API/build files from video_engine/. BUG=2535 R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/direct_transport.cc
|