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gerrit-public.fairphone.software / platform / external / webrtc / b351d6a8d8ded3fa114b40cefdff7ae12a5fdb4f
  1. b351d6a Reverting rev 929 due to failing assert on Linux. by stefan@webrtc.org · 13 years ago
  2. 9b18ed6 Removed incorrect dependency. by phoglund@webrtc.org · 13 years ago
  3. fd3a0ef RTP bw estimate fix. by mflodman@webrtc.org · 13 years ago
  4. 1144ba2 Base and codec tests now run verify output and render to file instead of to screen. by phoglund@webrtc.org · 13 years ago
  5. 62e48eb adding owners for test by niklas.enbom@webrtc.org · 13 years ago
  6. 50b3cbe First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable. by niklas.enbom@webrtc.org · 13 years ago
  7. b61c410 Fixed a couple of Android makefiles to let voe and vie build properly. by kma@webrtc.org · 13 years ago
  8. 13318ef (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing. by kma@webrtc.org · 13 years ago
  9. 7a4eb28 Calculate the available bandwidth before sending a TMMBR by mflodman@webrtc.org · 13 years ago
  10. 637a59e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. by mflodman@webrtc.org · 13 years ago
  11. 855a77c Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo. by tina.legrand@webrtc.org · 13 years ago
  12. c4f129f Improve the mixing saturation protection scheme. by andrew@webrtc.org · 13 years ago
  13. 41f3855 Upgrade libvpx to b615a6d4. by marpan@webrtc.org · 13 years ago
  14. d30b688 Remove TraceScan executable. by andrew@webrtc.org · 13 years ago
  15. 4b13fc9 Add delay modification to process_test. by andrew@webrtc.org · 13 years ago
  16. 2f32b5c Fixes an issue where file playing could happen at a lower sampling frequency than the file. by henrike@webrtc.org · 13 years ago
  17. eb4ef17 Removing vplib include and VideoInterpolator when not needed by mikhal@webrtc.org · 13 years ago
  18. 488ed92 Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  19. c3a4dcd Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  20. ad79d6f Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
  21. 03a9eb1 RTP module: Make sure payloadName is null terminated. by mflodman@webrtc.org · 13 years ago
  22. f3c1b87 my eyes started bleeding when I saw this... by niklas.enbom@webrtc.org · 13 years ago
  23. 4d8cd9d Adding GetOutputDir method to test_support library. by kjellander@webrtc.org · 13 years ago
  24. 9dcab8f Restoring Android.mk by kjellander@webrtc.org · 13 years ago
  25. 4cd841e Fix win compile error for interpolator_test by niklas.enbom@webrtc.org · 13 years ago
  26. cff98ca Made it possible to run the voe_auto_test standard test in GTest behind a flag. The purpose is to run the whole test without any manual intervention since we want to run the test on a build bot in automated mode. by phoglund@webrtc.org · 13 years ago
  27. c58ef08 Removes system CPU measurement for Chrome build. by henrikg@webrtc.org · 13 years ago
  28. f15fbc3 Change in RTP module SendVP8 by henrik.lundin@webrtc.org · 13 years ago
  29. 9b81351 Changes for building audio coding in anroid. Only makefiles are touched. by kma@webrtc.org · 13 years ago
  30. 26d3667 Fix for broken test after r897 by henrike@webrtc.org · 13 years ago
  31. e2a34f8 Removes the API for setting RX VAD since the RX vad should always be on anyways. by henrike@webrtc.org · 13 years ago
  32. 5ae9f5e Adding logs in RTPSender::ReSendToNetwork. by mflodman@webrtc.org · 13 years ago
  33. bf48384 Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files. by kjellander@webrtc.org · 13 years ago
  34. 36e1ad9 Restructuring and removing ilbc_test.gypi. by kjellander@webrtc.org · 13 years ago
  35. 689cb30 First version of PythonCharts. by kjellander@webrtc.org · 13 years ago
  36. b353d21 ...and now fix the Debug build. by andrew@webrtc.org · 13 years ago
  37. 369766e Fix Release mode errors in common_video tests. by andrew@webrtc.org · 13 years ago
  38. a5c4c1f Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor. by vikasmarwaha@webrtc.org · 13 years ago
  39. 040cb71 Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT. by marpan@webrtc.org · 13 years ago
  40. 2256269 Enabling all common_video targets in webrtc.gyp. by andrew@webrtc.org · 13 years ago
  41. 731e9ae Fixes ACM API test to build on 32-bits machines. by tina.legrand@webrtc.org · 13 years ago
  42. e2d2801 Update .gitignore. by andrew@webrtc.org · 13 years ago
  43. 20a370e Changing the namespace of TestSuite to webrtc::test. by kjellander@webrtc.org · 13 years ago
  44. 1a8d08a Changing usage of gtest_main target, to use test_support_main instead. by kjellander@webrtc.org · 13 years ago
  45. 89088b9 Fix the path to protoc.gypi. by andrew@webrtc.org · 13 years ago
  46. 2475a19 Committing a file that was part of CL 175002, but for wome reason weren't uploaded correctly. by tina.legrand@webrtc.org · 13 years ago
  47. fb389e3 This CL is divided in several patches, to make review easier. by tina.legrand@webrtc.org · 13 years ago
  48. a4b9660 Add mistakenly removed VAD enabling function. by andrew@webrtc.org · 13 years ago
  49. e203de7 jitter_buffer updates: by mikhal@webrtc.org · 13 years ago
  50. 7232ad7 reverted back the sanity and changed the test by pwestin@webrtc.org · 13 years ago
  51. cfc1070 Fixed sanity for min length by pwestin@webrtc.org · 13 years ago
  52. 075e91f Added parsing of width and height from VP8 header by pwestin@webrtc.org · 13 years ago
  53. 679cb07 Fix build error for release build by henrik.lundin@webrtc.org · 13 years ago
  54. baf6db5 Making dual decoder work again in VCM by henrik.lundin@webrtc.org · 13 years ago
  55. 4bb1410 A change to Android makefile for building voe auto test. by kma@webrtc.org · 13 years ago
  56. d292b9c Unit tests now compile and run at all platforms. Cosmetic changes to mocks.h. by kjellander@webrtc.org · 13 years ago
  57. 87c50f0 Adding author by niklas.enbom@webrtc.org · 13 years ago
  58. 3a9680b Adding author by niklas.enbom@webrtc.org · 13 years ago
  59. 0ba3133 Aligning license file with file header by niklas.enbom@webrtc.org · 13 years ago
  60. 895870b Adding marker bit to RTPanalyze results by henrik.lundin@webrtc.org · 13 years ago
  61. bb8dfbd updating vpm unit_test following r858 by mikhal@webrtc.org · 13 years ago
  62. 7395d3d Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 by turaj@webrtc.org · 13 years ago
  63. fac5316 Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn. by turaj@webrtc.org · 13 years ago
  64. 9116cf7 Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error. by turaj@webrtc.org · 13 years ago
  65. 29d75b3 Only allow increasing capture time. by mflodman@webrtc.org · 13 years ago
  66. 18ee6ec Use __inline in NS-fixed. by andrew@webrtc.org · 13 years ago
  67. 3119ecf Fix audioproc build errors on Windows. by andrew@webrtc.org · 13 years ago
  68. c4ab870 video_processing: Adding logic to avoid a memcpy when not required by mikhal@webrtc.org · 13 years ago
  69. 0ab521f Resolving a crash related to strncopy followed by a strcat by punyabrata@webrtc.org · 13 years ago
  70. 36a992b Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. by perkj@webrtc.org · 13 years ago
  71. d683770 Fixing VPMUnitTest compilation error on Windows. by kjellander@webrtc.org · 13 years ago
  72. b37c628 Fixes crash due to r841. Review URL: http://webrtc-codereview.appspot.com/256004 by henrike@webrtc.org · 13 years ago
  73. e9f909b Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. by kma@webrtc.org · 13 years ago
  74. f1a45d7 Add missing <stdlib.h> to data_log test. by andrew@webrtc.org · 13 years ago
  75. 3134aac Use fileutils for the audio file in voe_auto_test. by andrew@webrtc.org · 13 years ago
  76. 2795750 Changed Android makefile to make the lastest video render code run. by kma@webrtc.org · 13 years ago
  77. 8473688 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  78. 8885d22 by henrike@webrtc.org · 13 years ago
  79. 1e10bb3 Remove global std::strings from fileutils. by andrew@webrtc.org · 13 years ago
  80. 2c74bab Remove unneeded assert and tracing. by andrew@webrtc.org · 13 years ago
  81. 299e2c9 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame by amyfong@webrtc.org · 13 years ago
  82. 4d8c818 The implementation before this change list keeps the ownership of memory that is used by peer connection instances in the peer connection manager. This means that if the peer connection manager is deleted before all peer connections it has created, these peer connections will be pointing to invalid memory. by henrike@webrtc.org · 13 years ago
  83. 177bb52 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  84. 066f9e5 Ray, please verify that this cl fixes the issue. Once the verification has been made, please review: by henrike@webrtc.org · 13 years ago
  85. 731ecba by henrike@webrtc.org · 13 years ago
  86. 1f6d740 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). by braveyao@webrtc.org · 13 years ago
  87. 88e0a34 Remove duplicated code. Review URL: http://webrtc-codereview.appspot.com/251001 by wu@webrtc.org · 13 years ago
  88. f960211 Fixes two jitter buffer bugs related to NACK. by stefan@webrtc.org · 13 years ago
  89. 35a12cd Fix comment. by perkj@webrtc.org · 13 years ago
  90. 8129752 Add refcount and scoped_refptr. by perkj@webrtc.org · 13 years ago
  91. 94cfde7 Removed scoped_refptr from libjingle.gyp by perkj@webrtc.org · 13 years ago
  92. 7e08613 Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. by perkj@webrtc.org · 13 years ago
  93. 250cd6f Added a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file. by bjornv@webrtc.org · 13 years ago
  94. eb65860 Reverts the workaround in r823 and solves a macro bug. by stefan@webrtc.org · 13 years ago
  95. 8b1f621 Updated gypi for tests to work on osx. by tina.legrand@webrtc.org · 13 years ago
  96. dfbebb9 Add a documented_interfaces watchlist. by andrew@webrtc.org · 13 years ago
  97. ca4666b vie wintest added hybrid protection mode by amyfong@webrtc.org · 13 years ago
  98. 1e7e60b Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was by amyfong@webrtc.org · 13 years ago
  99. 51e1bb4 vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll by amyfong@webrtc.org · 13 years ago
  100. 5200a05 video_coding/jitter_buffer Updating condition on which we return a frame. by mikhal@webrtc.org · 13 years ago