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gerrit-public.fairphone.software / platform / external / webrtc / 8d27a1c7238fc66fad6b1928b7f92947eb0bfc38
  1. 8d27a1c Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
  2. 6879c8a Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
  3. 5c280ec Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
  4. 5fcddf2 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
  5. 390fcb7 Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. by henrike@webrtc.org · 11 years ago
  6. 28654cb Update talk folder to revision=49713299. by henrike@webrtc.org · 11 years ago
  7. 5bb8e7e Adjusted net delay perf expectations slightly. by phoglund@webrtc.org · 11 years ago
  8. 129afc2 Correctly rebuild WebRTCDemo after jni/ source file changes by yujie.mao@webrtc.org · 11 years ago
  9. dd02935 Roll libvpx to 211873. -pickup public roll to: 33149cbb by marpan@webrtc.org · 11 years ago
  10. 5e44b8f Add libjingle's valgrind suppressions by henrike@webrtc.org · 11 years ago
  11. 0df5b8d Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
  12. 4e4bf4d Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. by henrike@webrtc.org · 11 years ago
  13. c6d5b50 AppRTCDemo: build fixes for iOS build in webrtc by fischman@webrtc.org · 11 years ago
  14. 9f07ea4 Roll tools/android 4235:4258, to pick up an x86 md5sum_bin binary by yujie.mao@webrtc.org · 11 years ago
  15. d2102af Undo libvpx include changes in r4348 to fix build. by tnakamura@webrtc.org · 11 years ago
  16. 8c73471 talk: DataChannel.java repeated contents. This removes the duplicate. by henrike@webrtc.org · 11 years ago
  17. 9de257d Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository. by henrike@webrtc.org · 11 years ago
  18. a3f3014 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
  19. 64e2cbf clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  20. aa4d96a Revert r4301 by tnakamura@webrtc.org · 11 years ago
  21. 7b2f955 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. by henrike@webrtc.org · 11 years ago
  22. 4258154 Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle. by henrike@webrtc.org · 11 years ago
  23. 3d8647f Include files from webrtc/.. paths in signal_processing/. by pbos@webrtc.org · 11 years ago
  24. 0c4e05a Include files from webrtc/.. paths in media_file/. by pbos@webrtc.org · 11 years ago
  25. 9b82dce Make sure first RTP packet counts as in-order. by pbos@webrtc.org · 11 years ago
  26. 2e10b8e Include files from webrtc/.. paths in bitrate_controller/. by pbos@webrtc.org · 11 years ago
  27. a440732 Include files from webrtc/.. paths in video_coding/. by pbos@webrtc.org · 11 years ago
  28. 4a44ea2 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" by elham@webrtc.org · 11 years ago
  29. 4888fd4 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" by elham@webrtc.org · 11 years ago
  30. b7eda43 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  31. 6f5707e Revert r4328 by elham@webrtc.org · 11 years ago
  32. 8543c1c Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
  33. ca35c19 Roll libvpx to 208227. -pick up libvpx roll to 93f88ab. by marpan@webrtc.org · 11 years ago
  34. df119c9 Remove dead video_capture for QuickTime. by pbos@webrtc.org · 11 years ago
  35. 723d683 Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository. by henrike@webrtc.org · 11 years ago
  36. a9b74ad Include files from webrtc/.. paths in video_capture/. by pbos@webrtc.org · 11 years ago
  37. 8b06200 Include files from webrtc/.. paths in utility/. by pbos@webrtc.org · 11 years ago
  38. 0ed57c5 Remove dead code testAPI.cc. by pbos@webrtc.org · 11 years ago
  39. 5aa3f1b Include files from webrtc/.. paths in video_render/. by pbos@webrtc.org · 11 years ago
  40. 5b10d8f Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
  41. ffe16bd trunk/talk: removes empty folders. by henrike@webrtc.org · 11 years ago
  42. 811269d Include files from webrtc/.. paths in audio_device/. by pbos@webrtc.org · 11 years ago
  43. db6e3f8 Fix root-relative includes for pacing/. by pbos@webrtc.org · 11 years ago
  44. e4736ee Fixes a crash when sending SR reports from a sender only module. by stefan@webrtc.org · 11 years ago
  45. aeba6e8 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. by braveyao@webrtc.org · 11 years ago
  46. 96edd56 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  47. 69215d8 Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
  48. adf23a5 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
  49. 717d147 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  50. 9de89a6 Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 11 years ago
  51. 452d853 Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
  52. 08933a5 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  53. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago
  54. 6aa6229 Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
  55. c79b929 Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
  56. fc496d9 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
  57. f3f1358 Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
  58. cab716c Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. by stefan@webrtc.org · 11 years ago
  59. f56d612 Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
  60. af8d5af Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
  61. 5fc4d34 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
  62. 1a7b9b9 Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
  63. e80a934 Added modules_unittests.isolate for ndk-apk builds. by henrike@webrtc.org · 11 years ago
  64. a950300b Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
  65. a2073af Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
  66. bd3eee3 Fixes broken gyp-condition. by henrike@webrtc.org · 11 years ago
  67. 34773d9 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  68. 1932fe1 Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
  69. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  70. d4d9480 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. by mcasas@webrtc.org · 11 years ago
  71. db7d82f Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
  72. caf2fcc Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  73. 21beaf9 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
  74. 0b8636a In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. by braveyao@webrtc.org · 11 years ago
  75. 1303af3 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. by henrike@webrtc.org · 11 years ago
  76. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  77. 45426ea In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
  78. f6f033f Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
  79. b1698ab Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
  80. ecd3c80 Add Magnus to root owners. by tommi@webrtc.org · 11 years ago
  81. c66aaaf Rename unit_test.{cc,h} under module_unittest. by pbos@webrtc.org · 11 years ago
  82. 510dfad Update myself in webrtc watchlist by yujie.mao@webrtc.org · 11 years ago
  83. 65a1f2c Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
  84. 504af45 Diff NTP and internal once in VideoCaptureImpl. by pbos@webrtc.org · 11 years ago
  85. 546c91d Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  86. d4803ce WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
  87. 90cc3b9 Android opengles renderer: add thread sync to swap frame and draw native. by braveyao@webrtc.org · 11 years ago
  88. 5616aba Suppress excessive logging in video_coding by hclam@chromium.org · 11 years ago
  89. 2a7fd53 Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
  90. 83cebb2 Removes unused main function that is poluting the build. by henrike@webrtc.org · 11 years ago
  91. 0021632 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
  92. 1d4a2d5 Move TickTime::QueryOsForTicks out-of-line by fischman@webrtc.org · 11 years ago
  93. 4cf1a8a Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. by stefan@webrtc.org · 11 years ago
  94. 7bcc7e3 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
  95. 2de80dd Fix unnamed-type-template-args warnings on clang. by pbos@webrtc.org · 11 years ago
  96. 3145a64 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
  97. e6168f5 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
  98. 1c986e7 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  99. a5fd2f1 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
  100. 892d750 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. by solenberg@webrtc.org · 11 years ago