1. 3e9ad26 Refactor iOS AppRTC parsing code. by tkchin@webrtc.org · 10 years ago
  2. a71bb60 Revert 7750 "Don't reset sequence number for a stream on deactiv..." by sprang@webrtc.org · 10 years ago
  3. 31f7a0e Don't reset sequence number for a stream on deactivate/reactivate. by sprang@webrtc.org · 10 years ago
  4. 2faf7ee Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" by perkj@webrtc.org · 10 years ago
  5. 58edb83 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. by glaznev@webrtc.org · 10 years ago
  6. 0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  7. dab5d92 Use mirror image for Android AppRTCDemo local preview. by glaznev@webrtc.org · 10 years ago
  8. 8562f23 OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
  9. 308e7ff Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." by kjellander@webrtc.org · 10 years ago
  10. 2751f2a This adds an Android apk for running tests on the Java layer of PeerConnection. by perkj@webrtc.org · 10 years ago
  11. 88d14f4 Remove expensive and unnecessary memory alloc for sending black frames on video by thorcarpenter@google.com · 10 years ago
  12. bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 10 years ago
  13. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  14. edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 10 years ago
  15. f58b455 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  16. 6f6ef72 Add DCHECK to ensure that NetEq's packet buffer is not empty by henrik.lundin@webrtc.org · 10 years ago
  17. 2176db3 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) by henrika@webrtc.org · 10 years ago
  18. 930e004 Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
  19. c72a22c Add preliminary empty file videoframefactory.cc by magjed@webrtc.org · 10 years ago
  20. 4ef22d1 Setting Opus FEC as default by minyue@webrtc.org · 10 years ago
  21. 4ec19e3 Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." by tommi@webrtc.org · 10 years ago
  22. 858dbbc cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  23. 6a782c2 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. by henrike@webrtc.org · 10 years ago
  24. a73d746 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." by magjed@webrtc.org · 10 years ago
  25. bbd8cad cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. by magjed@webrtc.org · 10 years ago
  26. ece3890 Report total bitrate for all streams in GetStats. by pbos@webrtc.org · 10 years ago
  27. 35c1ace Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." by magjed@webrtc.org · 10 years ago
  28. a1f5b96 Remove unnecessary copying of libjingle resource files. by kjellander@webrtc.org · 10 years ago
  29. 52da44b WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution by magjed@webrtc.org · 10 years ago
  30. 312614a Add jmi field for packets discarded due to network error by guoweis@webrtc.org · 10 years ago
  31. 6ca6190 Fix a SCTP message reordering issue in datachannel.cc. by jiayl@webrtc.org · 10 years ago
  32. 8038d42 Follow-up fixes for G722 by henrik.lundin@webrtc.org · 10 years ago
  33. c492231 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. by henrike@webrtc.org · 10 years ago
  34. d819803 Wire up DSCP support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  35. 957e802 Refactor SetDefaultEncoderConfig to work on existing codecs. by pbos@webrtc.org · 10 years ago
  36. 3c1970f (Auto)update libjingle 79414100-> 79428003 by buildbot@webrtc.org · 10 years ago
  37. 188d3b2 Enable VP9 video codec support on webrtcvideoengine behind a field trial. by andresp@webrtc.org · 10 years ago
  38. f85dbce Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" by henrik.lundin@webrtc.org · 10 years ago
  39. d105cc8 Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 10 years ago
  40. a2ef4fe Prevent a lot of VideoSendStream reconfigures. by pbos@webrtc.org · 10 years ago
  41. 82775b1 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. by andresp@webrtc.org · 10 years ago
  42. 5e16066 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). by henrika@webrtc.org · 10 years ago
  43. dced5d7 Revert "Advertise G722 as 8 kHz rather than 16 kHz" by henrik.lundin@webrtc.org · 10 years ago
  44. 34bda43 (Auto)update libjingle 79326895-> 79329222 by buildbot@webrtc.org · 10 years ago
  45. e5421e9 Volume buttons in AppRTCDemo should affect output audio volume. by henrika@webrtc.org · 10 years ago
  46. fd0efb6 Remove deprecated PeerConnection APIs. by perkj@webrtc.org · 10 years ago
  47. 19b4741 Removing unused method GetDefaultVideoEncoderConfig. by andresp@webrtc.org · 10 years ago
  48. 0ef890a (Auto)update libjingle 79285346-> 79320771 by buildbot@webrtc.org · 10 years ago
  49. 6340acd AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. by mcasas@webrtc.org · 10 years ago
  50. 1dcca40 Advertise G722 as 8 kHz rather than 16 kHz by henrik.lundin@webrtc.org · 10 years ago
  51. ee9d61c This fixes a small memory leak (found using Xcode/Instruments on iOS) in by tkchin@webrtc.org · 10 years ago
  52. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  53. a22a628 (Auto)update libjingle 79205306-> 79244016 by buildbot@webrtc.org · 10 years ago
  54. 795d003 (Auto)update libjingle 79200114-> 79205306 by buildbot@webrtc.org · 10 years ago
  55. 8125744 Cleanup RTCVideoRenderer interface. by tkchin@webrtc.org · 10 years ago
  56. 45ecf4c (Auto)update libjingle 79169148-> 79192489 by buildbot@webrtc.org · 10 years ago
  57. 8944c9d AppRTCDemoActivity: use differnet Themes for different API levels by mcasas@webrtc.org · 10 years ago
  58. fad9aec Remove protected files from talk/PRESUBMIT.py. by pbos@webrtc.org · 10 years ago
  59. 88ef632 Falling back on single-stream on multiple SSRC. by pbos@webrtc.org · 10 years ago
  60. b5d045e ReAdd PeerConnectionInterface::AddStream to fix Chrome build. by perkj@webrtc.org · 10 years ago
  61. 18de6f9 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. by tommi@webrtc.org · 10 years ago
  62. c2dd5ee Prepare for removal of PeerConnectionObserver::OnError. by perkj@webrtc.org · 10 years ago
  63. a663d90 (Auto)update libjingle 79104430-> 79104922 by buildbot@webrtc.org · 10 years ago
  64. 5f38c8d Android AppRTCDemo improvements: by glaznev@webrtc.org · 10 years ago
  65. 96a9325 Implement external decoder support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  66. 2236267 Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan by henrik.lundin@webrtc.org · 10 years ago
  67. 5072e0f Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
  68. c2c94a9 Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 by kjellander@webrtc.org · 10 years ago
  69. 78c222b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  70. 8a130c1 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
  71. b7ed779 Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  72. 3bf3d23 Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  73. 2dc6f31 Adapting bitrate according to maxplaybackrate for Opus. by minyue@webrtc.org · 10 years ago
  74. 14146e4 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
  75. 50ca986 Improve the logging when a TCP connection is deleted. by jiayl@webrtc.org · 10 years ago
  76. 8219529 Cleaning up r7562-7567. by minyue@webrtc.org · 10 years ago
  77. 879fac8 (Auto)update libjingle 78822708-> 78823675 by buildbot@webrtc.org · 10 years ago
  78. 5f73a37 Revert 7563 "before rebase" due to wrong submission by minyue@webrtc.org · 10 years ago
  79. c11cc8d Revert 7564 "to submit" due to wrong submission by minyue@webrtc.org · 10 years ago
  80. de386bf to submit by minyue@webrtc.org · 10 years ago
  81. c673bb9 before rebase by minyue@webrtc.org · 10 years ago
  82. 0b62672 adding default rates by minyue@webrtc.org · 10 years ago
  83. 776e6f2 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  84. 1abc146 (Auto)update libjingle 78738075-> 78738103 by buildbot@webrtc.org · 10 years ago
  85. 7998089 ApprtDemo Android: Switch between front and back camera. by perkj@webrtc.org · 10 years ago
  86. 2623695 Renaming bandwidth to bitrate in webrtcvoiceengine. by minyue@webrtc.org · 10 years ago
  87. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  88. ae694ef (Auto)update libjingle 78642371-> 78680406 by buildbot@webrtc.org · 10 years ago
  89. fbd55cb (Auto)update libjingle 78616359-> 78642371 by buildbot@webrtc.org · 10 years ago
  90. f15dee6 Check if a datachannel in the current local description is an sctp channel before assuming rtp. by tommi@webrtc.org · 10 years ago
  91. 243eb8e Adding setting screen to AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  92. 068b529 (Auto)update libjingle 78583324-> 78583691 by buildbot@webrtc.org · 10 years ago
  93. 2e7ee4b Fix the SrtpFilter crash caused by two local offers. by pthatcher@webrtc.org · 10 years ago
  94. efc82c2 Implement screencast settings for WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  95. 1732df6 Use flags set by the port allocator. by braveyao@webrtc.org · 10 years ago
  96. 3f7bcc1 (Auto)update libjingle 78430441-> 78445452 by buildbot@webrtc.org · 10 years ago
  97. c7ed8db (Auto)update libjingle 78427027-> 78430441 by buildbot@webrtc.org · 10 years ago
  98. 4709887 Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. by perkj@webrtc.org · 10 years ago
  99. c9d6d14 patch from issue 25469004 by pthatcher@webrtc.org · 10 years ago
  100. 8fe75ee (Auto)update libjingle 78381351-> 78389679 by buildbot@webrtc.org · 10 years ago