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gerrit-public.fairphone.software / platform / external / webrtc / cb858ba3974c921627e81805e2ab4a2ae52c6619
  1. cb858ba Make an AudioEncoder subclass for iLBC by kwiberg@webrtc.org · 10 years ago
  2. ee43263 Cleaned up real_fft APIs due to non-existing NEON code by bjornv@webrtc.org · 10 years ago
  3. ed7824b Change Android PeerConnectionUnittest to build using Chrome macros. by perkj@webrtc.org · 10 years ago
  4. ba8138b Change type of nack_last_time_sent_full_ from uint32_t to int64_t. by asapersson@webrtc.org · 10 years ago
  5. aefe61a PRESUBMIT: Add check for checkdeps. by kjellander@webrtc.org · 10 years ago
  6. 7db359b Roll chromium_revision 24b4c73..8e72e1d by kjellander@webrtc.org · 10 years ago
  7. d91d359 PRESUBMIT: Add iOS ARM64 trybots to default set. by kjellander@webrtc.org · 10 years ago
  8. fb01376 Adjust some parameters for VP9 tests. by marpan@webrtc.org · 10 years ago
  9. e2a9261 Improve AppRTCDemo connection speed by sending all by glaznev@webrtc.org · 10 years ago
  10. bd8cc0b Add codereview.settings to the /talk subdirectory by kjellander@webrtc.org · 10 years ago
  11. 5af8cd7 Add codereview.settings to the /webrtc subdirectory by kjellander@webrtc.org · 10 years ago
  12. 599e299 cricket::VideoFrame int64 to int64_t. by kjellander@webrtc.org · 10 years ago
  13. 9b5467e Fix assertion failure when closing data channel, and add a unit test. by bemasc@webrtc.org · 10 years ago
  14. 4b407aa Update AppRTCDemo README with information on 3-dot-apprtc server by glaznev@webrtc.org · 10 years ago
  15. 7169afd With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. by guoweis@webrtc.org · 10 years ago
  16. 369746b Support new WebSocket signaling format. by glaznev@webrtc.org · 10 years ago
  17. 0b38478 Add support for parsing header only RTP dumps with bwe_rtp_play. by stefan@webrtc.org · 10 years ago
  18. 9f79fe6 Merge remote bitrate estimator changes. by pbos@webrtc.org · 10 years ago
  19. 33ccdfa Relanding r7807. by minyue@webrtc.org · 10 years ago
  20. 52bc4f4 Revert 7807 "Removing unused opus wrapper APIs." by minyue@webrtc.org · 10 years ago
  21. c0991fe Roll chromium_revision 24b4c73..f27c369 by kjellander@webrtc.org · 10 years ago
  22. e54a634 Removing unused opus wrapper APIs. by minyue@webrtc.org · 10 years ago
  23. 8c9ff20 Redo the change of https://webrtc-codereview.appspot.com/30949004/ by guoweis@webrtc.org · 10 years ago
  24. fd84229 Revert "Implement GetState() for channel's connectivity check state." by guoweis@webrtc.org · 10 years ago
  25. ff72f9e Implement GetState() for channel's connectivity check state. by guoweis@webrtc.org · 10 years ago
  26. fd4acf6 Adding WebRtcSpl_MaxAbsValueW16 intrinsics version by andrew@webrtc.org · 10 years ago
  27. 3a52458 add WebRtcIsacfix_AutocorrNeon's intrinsics version by andrew@webrtc.org · 10 years ago
  28. 8dc21dc Rename internal AudioEncoder::Encode method to EncodeInternal by henrik.lundin@webrtc.org · 10 years ago
  29. d1fac61 Remove need for assembly offset generation in aecm and ns module. by andrew@webrtc.org · 10 years ago
  30. 3800e13 Revert r7798 ("Move the AudioDecoder interface out of NetEq") by kwiberg@webrtc.org · 10 years ago
  31. 00ba1a7 Move the AudioDecoder interface out of NetEq by kwiberg@webrtc.org · 10 years ago
  32. 0fb6ad2 Check if cpu_monitor_ exists before Stop(). by pbos@webrtc.org · 10 years ago
  33. fa914e2 Adding a duration printout to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  34. d8aed6b Verify that cpu_monitor exists before calling Stop(). by asapersson@webrtc.org · 10 years ago
  35. c3e097c Add Android test runner script for WebRTC. by kjellander@webrtc.org · 10 years ago
  36. 8e5c814 Convert DEPS to only reference Git repos by kjellander@webrtc.org · 10 years ago
  37. 511f8a8 TurnPort should ignore STUN binding reponses when using shared socket. by jiayl@webrtc.org · 10 years ago
  38. 001f3b9 Adjust parameter in videoprocessor_integration_test for vp9. by marpan@webrtc.org · 10 years ago
  39. a7384a1 Simplify audio_buffer APIs by aluebs@webrtc.org · 10 years ago
  40. ceca014 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. by marpan@webrtc.org · 10 years ago
  41. eb09542 Don't reset sequence number for a stream on deactivate/reactivate. by pthatcher@webrtc.org · 10 years ago
  42. d019551 Change minimum video encoder initialization resolution to by glaznev@webrtc.org · 10 years ago
  43. 1751ee7 Remove -flax-vector-conversions flag for ARM NEON building. by andrew@webrtc.org · 10 years ago
  44. ac68ef9 Clear 2 unused functions in audio processing aecm module. by andrew@webrtc.org · 10 years ago
  45. beee9ce Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. by perkj@webrtc.org · 10 years ago
  46. 7f1dfa5 Adding a payload type to AudioEncoder objects by henrik.lundin@webrtc.org · 10 years ago
  47. 0cd5558 AudioEncoder subclass for G722 by kwiberg@webrtc.org · 10 years ago
  48. 84515f8 Roll chromium_revision 309cf65..24b4c73 by kjellander@webrtc.org · 10 years ago
  49. 5950b64 Use c++11 features in webrtc/base/network.cc as a test to see if we can use them. by pthatcher@webrtc.org · 10 years ago
  50. 146e0fd Fix the build by putting in a typecast to avoid a comparison between by pthatcher@webrtc.org · 10 years ago
  51. dea5173 Add start bitrate and vp8 hw acceleration option to Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  52. 32ec0dd (Auto)update libjingle 81063831-> 81073932 by buildbot@webrtc.org · 10 years ago
  53. 7f72249 Set simulcastIdx field to zero even if it has no meaning. by andresp@webrtc.org · 10 years ago
  54. 273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  55. 1db20a4 Adding EncodedInfo struct to AudioEncoder::Encode by henrik.lundin@webrtc.org · 10 years ago
  56. 20446e7 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat by henrik.lundin@webrtc.org · 10 years ago
  57. c93437e Add test NetEqDecodingTest.CngFirst by henrik.lundin@webrtc.org · 10 years ago
  58. 8331714 Adding a new test helper RtpFileWriter and use it in RTPcat by henrik.lundin@webrtc.org · 10 years ago
  59. 4796301 Whitespace change to force builds. by kjellander@webrtc.org · 10 years ago
  60. e75f2ce Add FORCE_HTTPS_COMMIT_URL to codereview.settings. by kjellander@webrtc.org · 10 years ago
  61. cc7755b Whitespace change by kjellander@webrtc.org · 10 years ago
  62. 74499ef Add whitespace.txt file. by kjellander@webrtc.org · 10 years ago
  63. 2c13f65 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. by tommi@webrtc.org · 10 years ago
  64. 83b5200 Add framerate for complete received frames to histogram stats: by asapersson@webrtc.org · 10 years ago
  65. cc144de Make bands vector in SplittingFilter Analysis const by aluebs@webrtc.org · 10 years ago
  66. 8789376 Move ChannelBuffer class to channel_buffer file by aluebs@webrtc.org · 10 years ago
  67. d87213a Remove unused RtpStatistics struct. by pbos@webrtc.org · 10 years ago
  68. 7d4e6d0 Roll chromium_revision d8c9041..309cf65 by kjellander@webrtc.org · 10 years ago
  69. d952c40 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  70. 3e9ad26 Refactor iOS AppRTC parsing code. by tkchin@webrtc.org · 10 years ago
  71. 79b9eba Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands by aluebs@webrtc.org · 10 years ago
  72. 7806d8f Fix an ASSERT that fires in a browser test for renegotiation. by jiayl@webrtc.org · 10 years ago
  73. a71bb60 Revert 7750 "Don't reset sequence number for a stream on deactiv..." by sprang@webrtc.org · 10 years ago
  74. a56a2c5 Enabling building with NEON on ARM64 by andrew@webrtc.org · 10 years ago
  75. 31f7a0e Don't reset sequence number for a stream on deactivate/reactivate. by sprang@webrtc.org · 10 years ago
  76. 91d928e Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
  77. 2faf7ee Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" by perkj@webrtc.org · 10 years ago
  78. 58edb83 Add video encoder fps and bitrate statistics to Android AppRTCDemo UI. by glaznev@webrtc.org · 10 years ago
  79. 0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  80. b951eb1 Add back EXPECT_TRUEs. by pbos@webrtc.org · 10 years ago
  81. ba25347 Reenable GetStats test. by pbos@webrtc.org · 10 years ago
  82. dab5d92 Use mirror image for Android AppRTCDemo local preview. by glaznev@webrtc.org · 10 years ago
  83. 03499a0 Add wav output capability to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  84. aff1751 Add new test for VP8 packetizer to test tight partitions by henrik.lundin@webrtc.org · 10 years ago
  85. dde19a6 sync_chromium.py: Check for chromium/src by kjellander@webrtc.org · 10 years ago
  86. 3398a4a PRESUBMIT: Only notify GN changes for GYP files in webrtc/* by kjellander@webrtc.org · 10 years ago
  87. 8562f23 OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
  88. 4f16c87 Simplifying VideoReceiver and JitterBuffer. by pbos@webrtc.org · 10 years ago
  89. 9334ac2 Use vector of CSRCs for DeliverFrame & SetCSRCs. by pbos@webrtc.org · 10 years ago
  90. 308e7ff Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." by kjellander@webrtc.org · 10 years ago
  91. 2751f2a This adds an Android apk for running tests on the Java layer of PeerConnection. by perkj@webrtc.org · 10 years ago
  92. 88d14f4 Remove expensive and unnecessary memory alloc for sending black frames on video by thorcarpenter@google.com · 10 years ago
  93. 1153322 Build fix for MIPS Android Webview build. by andrew@webrtc.org · 10 years ago
  94. bdcf38c cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class by magjed@webrtc.org · 10 years ago
  95. ad0e71c Update mock_frame_dropper.h to use size_t by kjellander@webrtc.org · 10 years ago
  96. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  97. edc6e57 Support loopback mode and command line execution by glaznev@webrtc.org · 10 years ago
  98. 6ff3ac1 Fix problems if first packet into NetEq is rejected by henrik.lundin@webrtc.org · 10 years ago
  99. ed91068 Create a NetEq test for when the first incoming payload type is unknown by henrik.lundin@webrtc.org · 10 years ago
  100. 049e4ec Change default values for CpuOveruseOptions. by asapersson@webrtc.org · 10 years ago