e74eef19bd3f101208dc72b98038e42fc523a351 |
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08-Jan-2016 |
stefan <stefan@webrtc.org> |
Add CreateSend/ReceiveTransport() methods to CallTest. This allows the test to create its own transports if it, for instance, needs to do demuxing. BUG=webrtc:5416 Review URL: https://codereview.webrtc.org/1573453002 Cr-Commit-Position: refs/heads/master@{#11187}
/external/webrtc/webrtc/test/call_test.h
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9fea80f50daab46f20d4a6fc67b0144fbbbf56cd |
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07-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Add audio streams to CallTest and a first A/V call test. Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers. Audio streams are using a fake audio device with file input. The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code. R=pbos@webrtc.org TBR=kjellander@webrtc.org BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1542653002 . Cr-Commit-Position: refs/heads/master@{#11171}
/external/webrtc/webrtc/test/call_test.h
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ff483617a4fdf282bb82d7f4ce15af3dbe305a4a |
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21-Dec-2015 |
stefan <stefan@webrtc.org> |
Step 1 to prepare call_test.* for combined audio/video tests. Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests. No functional changes. BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1537273003 Cr-Commit-Position: refs/heads/master@{#11101}
/external/webrtc/webrtc/test/call_test.h
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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/call_test.h
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1295297153ff0487580faf821f24f09a7c16ce30 |
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29-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Register header extensions in RtpRtcpObserver to avoid log spam. BUG=webrtc:5118 R=pbos@webrtc.org Review URL: https://codereview.webrtc.org/1416783006 . Cr-Commit-Position: refs/heads/master@{#10450}
/external/webrtc/webrtc/test/call_test.h
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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/call_test.h
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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/call_test.h
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4f4ec0a9270a8cefadfa12e9fa3b979b58b15392 |
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22-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Re-Land: Implement AudioReceiveStream::GetStats(). R=tommi@webrtc.org BUG=webrtc:4690 Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0 Review URL: https://codereview.webrtc.org/1390753002 . Cr-Commit-Position: refs/heads/master@{#10369}
/external/webrtc/webrtc/test/call_test.h
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2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
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28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces. BUG=webrtc:1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1369263002 Cr-Commit-Position: refs/heads/master@{#10096}
/external/webrtc/webrtc/test/call_test.h
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4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d |
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28-Aug-2015 |
solenberg <solenberg@webrtc.org> |
Add send transports to individual webrtc::Call streams. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1273363005 Cr-Commit-Position: refs/heads/master@{#9807}
/external/webrtc/webrtc/test/call_test.h
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e62202fedf57b74cc263246c0586ee353978caf8 |
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21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8. This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
/external/webrtc/webrtc/test/call_test.h
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9526187dde1e93389b1d9077287eade974f9acfb |
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10-Apr-2015 |
Erik Språng <sprang@google.com> |
Default enable abs send time bwe for CallTest Using the single stream bwe is really bad for the screenshare test case in particular, but would probably help in other cases as well so enabling it by default in CallTest setup. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43089004 Cr-Commit-Position: refs/heads/master@{#8971}
/external/webrtc/webrtc/test/call_test.h
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14665ff7d4024d07e58622f498b23fd980001871 |
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04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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00b8f6b3643332cce1ee711715f7fbb824d793ca |
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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/call_test.h
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8f27fcce79584378da97f0d84574564799e138d6 |
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09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..." Reasons for revert: 1. glaznev discovered potentially related problems using the Android AppRTCDemo. 2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky. > Support associated payload type when registering Rtx payload type. > > Major changes include, > - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. > - Receiver: Restore RTP packets by the new RTX-APT map. > - Sender: Send RTP packets by checking RTX-APT map. > - Add RTX payload type for RED in the default codec list. > > BUG=4024 > R=pbos@webrtc.org, stefan@webrtc.org > TBR=mflodman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26259004 > > Patch from Changbin Shao <changbin.shao@intel.com>. TBR=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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2a169640a3225a559f926fe74f1fe1af239e191f |
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09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type. Major changes include, - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. - Receiver: Restore RTP packets by the new RTX-APT map. - Sender: Send RTP packets by checking RTX-APT map. - Add RTX payload type for RED in the default codec list. BUG=4024 R=pbos@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26259004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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776e6f289c7396a1143b8b36b03f88b08ac8cba3 |
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29-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use external VideoDecoders in VideoReceiveStream. Removes direct VideoCodec use from the new API, exposes VideoDecoders through webrtc/video_decoder.h similar to VideoEncoders. Also includes some preparation for wiring up external decoders in WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they were allocated internally or externally. Additionally addresses a data race in VideoReceiver that was exposed with this change. R=mflodman@webrtc.org, stefan@webrtc.org TBR=pthatcher@webrtc.org BUG=2854,1667 Review URL: https://webrtc-codereview.appspot.com/27829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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bbe0a8517d7f9da7aa779bff77cdbb70df358437 |
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19-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Config struct for VideoEncoder. Used for config parameters in common between multiple codecs as well as the encoder-specific pointer. In particular this contains content mode (realtime video vs. screenshare). BUG=1788 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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01581da71145d4b9504d12cfad0c988d1fc68654 |
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04-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Fix audio/video sync when FEC is enabled. Also improves the tests by adding a test case for FEC, and running the a/v sync tests with NACK and simulated packet loss. BUG=crbug/374104 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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7ae9108b60cef53444931ce794869d303ab8f9ab |
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10-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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91f1752f2d34eee653f7693e09a485a8f5c50e1e |
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10-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support VP8 encoder settings in VideoSendStream. Stop-gap solution to support VP8 codec settings in the new API until encoder settings can be passed on to the VideoEncoder without requiring explicit support for the codec. BUG=3424 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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2bb1bdab8d11f5445693c028335fb3ace631f636 |
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07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams. A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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be9d2a45499d87f3b04e644fc173b0d997a9eeea |
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30-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reserve RTP/RTCP modules in SetSSRC. Allows setting SSRCs for future simulcast layers even though no set send codec uses them. Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required for bitrate ramp-up, instead of send-side only (resolving issue 3078). This test was used to verify reserved modules' SSRCs are preserved correctly. To enable a multiple-stream end-to-end test test::CallTest was modified to work on a vector of receive streams instead of just one. BUG=3078 R=kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15859005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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994d0b7229a18b255d81979c2bedaf8ecfae9bd7 |
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27-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor Call-based tests. Greatly reduces duplication of constants and setup code for tests based on the new webrtc::Call APIs. It also makes it significantly easier to convert sender-only to end-to-end tests as they share more code. BUG=3035 R=kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/call_test.h
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