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gerrit-public.fairphone.software / platform / external / webrtc / 6a34d584b8fcae89f8177fa4ca1c92efc5e3b739
  1. 6a34d58 Implement MediaStreamProxy. by perkj@webrtc.org · 13 years ago
  2. 4c059d8 Add metric for number of packets discarded by JB due to not being decodable by stefan@webrtc.org · 13 years ago
  3. 77d7d54 Replace the DestroyDeviceInfo with a virtual destructor. by wu@webrtc.org · 13 years ago
  4. 38e400a Adding native client test page to test loopback. by perkj@webrtc.org · 13 years ago
  5. e5542a0 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) by amyfong@webrtc.org · 13 years ago
  6. 6330cf2 Fixed ViE AutoTest trace file names to be consistent by amyfong@webrtc.org · 13 years ago
  7. ea89922 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. by wu@webrtc.org · 13 years ago
  8. 199f4de Rename all .cc files which include Objective-C headers to .mm. by andrew@webrtc.org · 13 years ago
  9. a0258de Fixes test build errors (warnings treated as errors) in system_wrappers. by henrike@webrtc.org · 13 years ago
  10. 26c9ff9 Add dummy implementation of DataLog::Combine method by henrik.lundin@webrtc.org · 13 years ago
  11. 791eec7 Add API to get the number of packets discarded by the video jitter buffer due to being too late. by stefan@webrtc.org · 13 years ago
  12. 06887ae Fixes two bugs when decoding with packet losses. by stefan@webrtc.org · 13 years ago
  13. ed081a9 Print info about the local and remote resolution in the Windows client. by tommi@webrtc.org · 13 years ago
  14. 73ba416 Fix OnClose(socket, NO_ERROR) compile error on Linux. by perkj@webrtc.org · 13 years ago
  15. 1843664 DataLog: Changing from common_types to typedefs by henrik.lundin@webrtc.org · 13 years ago
  16. f7b36a4 Fix bug in the server where a wait request was incorrectly handled. by tommi@webrtc.org · 13 years ago
  17. c0b2250 Fix the Windows build. Review URL: http://webrtc-codereview.appspot.com/213004 by tommi@webrtc.org · 13 years ago
  18. 5a695d6 Fix bug in the client that caused signaling messages to be dropped. by tommi@webrtc.org · 13 years ago
  19. d855bd4 C wrapper for DataLog class by henrik.lundin@webrtc.org · 13 years ago
  20. 6364d12 Fix a couple of build warnings. by tommi@webrtc.org · 13 years ago
  21. e95458c Started rewriting video_engine tests to use GUnit. by phoglund@webrtc.org · 13 years ago
  22. c8c4deb Fix Windows build. %zu isn't supported in the crt implementation by tommi@webrtc.org · 13 years ago
  23. 5a945ec A little upgrade to the HTML test page: by tommi@webrtc.org · 13 years ago
  24. 25e0b8e Python output flag and keyframe interval flags. by kjellander@webrtc.org · 13 years ago
  25. a31b254 Python output flag and keyframe interval flags. by kjellander@webrtc.org · 13 years ago
  26. 80dd19b vplib tests: Removing old and unused file and directories. by mikhal@webrtc.org · 13 years ago
  27. f6ab63c Update PeerConnection_client to open a video capture device. by perkj@webrtc.org · 13 years ago
  28. bf54ef9 Removed code under a non-existing define. by henrike@webrtc.org · 13 years ago
  29. 1a2933c Fixes a Valgrind warning triggering when the number of pending messages hit the limit. by henrike@webrtc.org · 13 years ago
  30. 2915f6f Use proper printf size_t specifier to fix Linux 32-bit build. by andrew@webrtc.org · 13 years ago
  31. b2d4921 Remove trailing whitespace in AudioDevice. by andrew@webrtc.org · 13 years ago
  32. d6132f5 by mikhal@webrtc.org · 13 years ago
  33. 3a6d4f4 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams. by perkj@webrtc.org · 13 years ago
  34. 35a1756 First version of video quality measurement program and test framework. by kjellander@webrtc.org · 13 years ago
  35. 3ce62fc Move merge_libs targets to their own gyp. by andrew@webrtc.org · 13 years ago
  36. af57de0 Some code style changes in audio_processing/ns/main/source/ by Astyle, by kma@webrtc.org · 13 years ago
  37. fa41d80 Fixes session state transition and registering observer. by mallinath@webrtc.org · 13 years ago
  38. 01ca01f Adding neteq_tests to modules tests by henrik.lundin@webrtc.org · 13 years ago
  39. 29787c7 Changes to WebRtcSession after Provider(s) interface addition. by mallinath@webrtc.org · 13 years ago
  40. bbc1f10 Changed modules/audio_processing/utility/Android.mk, to correct a build error in by kma@webrtc.org · 13 years ago
  41. 487e401 Moving creation of sessiondescriptions to webrtcsession. by perkj@webrtc.org · 13 years ago
  42. bf39ff4 Some general optimization in NS. by kma@webrtc.org · 13 years ago
  43. a58224f Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7. by kma@webrtc.org · 13 years ago
  44. cb4ab65 Moved creation of objects to the signaling thread. by perkj@webrtc.org · 13 years ago
  45. bafca10 Temp hook in WebRtcSession to VideoChannel. by mallinath@webrtc.org · 13 years ago
  46. 4b6f747 Fixes a newly introduced bug in the jitter buffer where buffer reallocation by stefan@webrtc.org · 13 years ago
  47. 93d216c Fixed bug in jitter buffer which caused the missingFrames bit to never be set. by stefan@webrtc.org · 13 years ago
  48. 61b4abf Proper use of frame rate argument in generic_codec_test. by stefan@webrtc.org · 13 years ago
  49. e06be4f video coding tests: Adding ssimFrame to interface by mikhal@webrtc.org · 13 years ago
  50. ae7a052 video_coding robustness: Updating hybrid mode's settings by mikhal@webrtc.org · 13 years ago
  51. 1b6ff7a Connecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. by perkj@webrtc.org · 13 years ago
  52. 666f56b MediaStreamHandler implements eventhandlers for streams and tracks. by perkj@webrtc.org · 13 years ago
  53. 236fcaa Interface changes after we have the Serialize and Deserialize. by wu@webrtc.org · 13 years ago
  54. ed6d555 * Add the crypto serialize and deserialize. * Populate candidates test data. by wu@webrtc.org · 13 years ago
  55. ee2c391 more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. by mallinath@webrtc.org · 13 years ago
  56. f1f3fb3 Update to rate-mismatch factor in media_opt_util. by marpan@google.com · 13 years ago
  57. 99239d5 First compiling version of peerconnection_client_dev using the new Peerconnection API. by perkj@webrtc.org · 13 years ago
  58. f458916 Returning errors if any of the Init() settings in VoE fail. by andrew@webrtc.org · 13 years ago
  59. 5b91464 Allow an aggregated partition to spill over to a new packet. by stefan@webrtc.org · 13 years ago
  60. 1ba3dbe Adds possibility to log delay estimates in AEC. by bjornv@google.com · 13 years ago
  61. f72c367 Reverting changelist 666 since it broke the build on Mac. by stefan@webrtc.org · 13 years ago
  62. 6d169f2 Fix Mac build error in vie_auto_test introduced in r666. by andrew@webrtc.org · 13 years ago
  63. c93e363 * Add Deserize for PeerConnectionMessage by wu@webrtc.org · 13 years ago
  64. e90265b Commit http://webrtc-codereview.appspot.com/191001/ by tommi@webrtc.org · 13 years ago
  65. e804ee1 This patch hooks up PeerConnectionImpl to PeerConnectionSignaling. by perkj@webrtc.org · 13 years ago
  66. 78083bf * Add Serialize functions to PeerConnectionMessage. by wu@webrtc.org · 13 years ago
  67. 9a1249d first cut of webrtcsession. Doesn't do much other than creating files and empty function bodies. by mallinath@webrtc.org · 13 years ago
  68. 5eec6cf Started rewriting video_engine tests to use GUnit. by mflodman@webrtc.org · 13 years ago
  69. 5045f67 Add SignalUpdateSessionDescription to PeerConnectionSignaling. by perkj@webrtc.org · 13 years ago
  70. 6b6d081 Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected. by punyabrata@webrtc.org · 13 years ago
  71. c611b1a Bit-exact with non-Neon version. by kma@google.com · 13 years ago
  72. 87d4979 Add patterns for root_files (src/build/ and non-recursive contents of ./ and src/), common_audio, and audio_processing to WATCHLISTS. by andrew@webrtc.org · 13 years ago
  73. 0beae67 Removed level estimator calls, since it is not supported. There are still one place left; used within SetRTPAudioLevelIndicationStatus(). The error return value of level_estimator() has no effect there. by bjornv@google.com · 13 years ago
  74. 2f56ff4 Implementation of PcSignaling. A Class to handle signaling between peerconnections. by perkj@webrtc.org · 13 years ago
  75. 18421f2 Remove unnecessary include from NS interface. by andrew@webrtc.org · 13 years ago
  76. 6a23ad5 Fixed the CameraCap button to say Version, also change the function name inside ChannelDlg.cpp by amyfong@webrtc.org · 13 years ago
  77. 2d08d43 * Added modification of Start Bit Rate to vie_auto_test_custom_call by amyfong@webrtc.org · 13 years ago
  78. 848fad2 video_coding: Updating media opt test - fixing call to protection callback. by mikhal@webrtc.org · 13 years ago
  79. 49d025f Get the right guid str for GetRecordingDeviceName by xians@google.com · 13 years ago
  80. 82f66a7 Return to the WebM git repository for libvpx. by andrew@webrtc.org · 13 years ago
  81. a2c6ea0 Removed a segmentation fault error when processing near_file only. by bjornv@google.com · 13 years ago
  82. 961885a In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7. by kma@google.com · 13 years ago
  83. e185e9f video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list. by mikhal@webrtc.org · 13 years ago
  84. cf13618 Deleting matlab files by turajs@google.com · 13 years ago
  85. 13335cc Deleting matlab files by turajs@google.com · 13 years ago
  86. 610f478 Deleting matlab files by turajs@google.com · 13 years ago
  87. 53439d9 Deleting matlab files by turajs@google.com · 13 years ago
  88. 713f91e Fixed vie_autotest_custom_call.cc minor issues. by amyfong@webrtc.org · 13 years ago
  89. 105ff39 video coding: updating offline tests. by mikhal@webrtc.org · 13 years ago
  90. 496ef8a To fix warnings in test files. by turajs@google.com · 13 years ago
  91. 8e9e83b This CL adds guards against division by zero, that should fix http://b/issue?id=5278531 by bjornv@google.com · 13 years ago
  92. 9e7774f Added compare methods for TickInterval class. by kjellander@webrtc.org · 13 years ago
  93. dca57bd Adding git ignore file. by kjellander@webrtc.org · 13 years ago
  94. dc743a8 Replaces a use of log2. by bjornv@google.com · 13 years ago
  95. 90eff6c Fix compilation error in build-in AEC test by leozwang@google.com · 13 years ago
  96. 221b522 Return the number of /dev/video* without trying to open it. by wu@webrtc.org · 13 years ago
  97. c389aa2 Fix the bad video issue on Window client by increasing the rtp recv buffer size. by ronghuawu@google.com · 13 years ago
  98. 65e6ab3 Temporary log2 remove to build in chrome by bjornv@google.com · 13 years ago
  99. 3be70ca Added mute, hold and typing detect to voe_cmd_test to increase functionality in the voe_cmd_test application. by amyfong@webrtc.org · 13 years ago
  100. a193042 When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. by wu@webrtc.org · 13 years ago