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webrtc / src / 776e6f289c7396a1143b8b36b03f88b08ac8cba3 / webrtc / video / rampup_tests.cc
  1. 32452b2 Make ReconfigureVideoEncoder use current bitrate. by pbos@webrtc.org · 10 years ago
  2. a73a678 Remove -1 from Call::Config::start_bitrate_bps. by pbos@webrtc.org · 10 years ago
  3. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 11 years ago
  4. 541753f Re-enable rampup_tests.cc for Android. by andresp@webrtc.org · 11 years ago
  5. 6f729e8 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 11 years ago
  6. f24c4a3 Fix flaky ramp-up test. by stefan@webrtc.org · 11 years ago
  7. b8e9e44 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 11 years ago
  8. 3d7da88 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 11 years ago
  9. 62bafae Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  10. 994d0b7 Refactor Call-based tests. by pbos@webrtc.org · 11 years ago
  11. eb16b81 Implements start bitrate for new video API. by mflodman@webrtc.org · 11 years ago
  12. cb254aa Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 11 years ago
  13. fbb567d Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 11 years ago
  14. 6ae48c6 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 11 years ago
  15. caba2d2 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 11 years ago
  16. de1429e Add thread annotations to Call API. by pbos@webrtc.org · 11 years ago
  17. a5c8d2c Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 11 years ago
  18. dc80bae Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 11 years ago
  19. 02e749f Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 11 years ago
  20. af839b2 Add AIMD option to BWE API. by stefan@webrtc.org · 11 years ago
  21. c148079 Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 11 years ago
  22. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  23. a714eaf Refactor rampup tests: by andresp@webrtc.org · 11 years ago
  24. 54464e6 Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 11 years ago
  25. b10363f3 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  26. 6ea4f63 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 11 years ago
  27. be39470 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  28. 1598b80 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  29. 36b6221 Adding a link to issue by henrik.lundin@webrtc.org · 11 years ago
  30. 998cb8f Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 11 years ago
  31. 845862f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 11 years ago
  32. d9b9560 Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
  33. c279a5d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  34. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  35. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  36. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  37. e9abd59 Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  38. 7e9315b Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  39. b613b5a Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  40. 2c46f8d Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
  41. 27326b6 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  42. ce90eff Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
  43. 53c8573 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
  44. 5a63655 Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
  45. 69969e2 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
  46. 16e03b7 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago[Renamed (94%) from webrtc/video_engine/test/rampup_tests.cc]
  47. ab65495 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 12 years ago
  48. 2902328 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
  49. 744fbc7 Split up EngineTests and RampupTests. by pbos@webrtc.org · 12 years ago