1. 89088b9 Fix the path to protoc.gypi. by andrew@webrtc.org · 13 years ago
  2. 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
  3. fb389e3 This CL is divided in several patches, to make review easier. by tina.legrand@webrtc.org · 13 years ago
  4. a4b9660 Add mistakenly removed VAD enabling function. by andrew@webrtc.org · 13 years ago
  5. e203de7 jitter_buffer updates: by mikhal@webrtc.org · 13 years ago
  6. 7232ad7 reverted back the sanity and changed the test by pwestin@webrtc.org · 13 years ago
  7. cfc1070 Fixed sanity for min length by pwestin@webrtc.org · 13 years ago
  8. 075e91f Added parsing of width and height from VP8 header by pwestin@webrtc.org · 13 years ago
  9. 679cb07 Fix build error for release build by henrik.lundin@webrtc.org · 13 years ago
  10. baf6db5 Making dual decoder work again in VCM by henrik.lundin@webrtc.org · 13 years ago
  11. 4bb1410 A change to Android makefile for building voe auto test. by kma@webrtc.org · 13 years ago
  12. d292b9c Unit tests now compile and run at all platforms. Cosmetic changes to mocks.h. by kjellander@webrtc.org · 13 years ago
  13. 87c50f0 Adding author by niklas.enbom@webrtc.org · 13 years ago
  14. 3a9680b Adding author by niklas.enbom@webrtc.org · 13 years ago
  15. 0ba3133 Aligning license file with file header by niklas.enbom@webrtc.org · 13 years ago
  16. 895870b Adding marker bit to RTPanalyze results by henrik.lundin@webrtc.org · 13 years ago
  17. bb8dfbd updating vpm unit_test following r858 by mikhal@webrtc.org · 13 years ago
  18. 7395d3d Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 by turaj@webrtc.org · 13 years ago
  19. 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
  20. 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
  21. 29d75b3 Only allow increasing capture time. by mflodman@webrtc.org · 13 years ago
  22. 18ee6ec Use __inline in NS-fixed. by andrew@webrtc.org · 13 years ago
  23. 3119ecf Fix audioproc build errors on Windows. by andrew@webrtc.org · 13 years ago
  24. c4ab870 video_processing: Adding logic to avoid a memcpy when not required by mikhal@webrtc.org · 13 years ago
  25. 0ab521f Resolving a crash related to strncopy followed by a strcat by punyabrata@webrtc.org · 13 years ago
  26. 36a992b Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. by perkj@webrtc.org · 13 years ago
  27. d683770 Fixing VPMUnitTest compilation error on Windows. by kjellander@webrtc.org · 13 years ago
  28. b37c628 Fixes crash due to r841. Review URL: http://webrtc-codereview.appspot.com/256004 by henrike@webrtc.org · 13 years ago
  29. e9f909b Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. by kma@webrtc.org · 13 years ago
  30. f1a45d7 Add missing <stdlib.h> to data_log test. by andrew@webrtc.org · 13 years ago
  31. 3134aac Use fileutils for the audio file in voe_auto_test. by andrew@webrtc.org · 13 years ago
  32. 2795750 Changed Android makefile to make the lastest video render code run. by kma@webrtc.org · 13 years ago
  33. 8473688 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  34. 8885d22 by henrike@webrtc.org · 13 years ago
  35. 1e10bb3 Remove global std::strings from fileutils. by andrew@webrtc.org · 13 years ago
  36. 2c74bab Remove unneeded assert and tracing. by andrew@webrtc.org · 13 years ago
  37. 299e2c9 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame by amyfong@webrtc.org · 13 years ago
  38. 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
  39. 177bb52 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  40. 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
  41. 731ecba by henrike@webrtc.org · 13 years ago
  42. 1f6d740 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). by braveyao@webrtc.org · 13 years ago
  43. 88e0a34 Remove duplicated code. Review URL: http://webrtc-codereview.appspot.com/251001 by wu@webrtc.org · 13 years ago
  44. f960211 Fixes two jitter buffer bugs related to NACK. by stefan@webrtc.org · 13 years ago
  45. 35a12cd Fix comment. by perkj@webrtc.org · 13 years ago
  46. 8129752 Add refcount and scoped_refptr. by perkj@webrtc.org · 13 years ago
  47. 94cfde7 Removed scoped_refptr from libjingle.gyp by perkj@webrtc.org · 13 years ago
  48. 7e08613 Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. by perkj@webrtc.org · 13 years ago
  49. 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
  50. eb65860 Reverts the workaround in r823 and solves a macro bug. by stefan@webrtc.org · 13 years ago
  51. 8b1f621 Updated gypi for tests to work on osx. by tina.legrand@webrtc.org · 13 years ago
  52. dfbebb9 Add a documented_interfaces watchlist. by andrew@webrtc.org · 13 years ago
  53. ca4666b vie wintest added hybrid protection mode by amyfong@webrtc.org · 13 years ago
  54. 1e7e60b Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was by amyfong@webrtc.org · 13 years ago
  55. 51e1bb4 vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll by amyfong@webrtc.org · 13 years ago
  56. 5200a05 video_coding/jitter_buffer Updating condition on which we return a frame. by mikhal@webrtc.org · 13 years ago
  57. 30f6376 VP8: Updating codec version: VP8 version will now return the libvpx version used. by mikhal@webrtc.org · 13 years ago
  58. 2d28aff Workaround for an issue where frames are grabbed for decoding prematurely. by stefan@webrtc.org · 13 years ago
  59. fbea4e5 Solves two bandwidth estimation issues and measures the sent video bitrate. by stefan@webrtc.org · 13 years ago
  60. 7e4269e Changed VP8 qp min and added noise reduction. by mflodman@webrtc.org · 13 years ago
  61. 8fc663b Don't trigger false ViE SetReceiveCodec warning. by mflodman@webrtc.org · 13 years ago
  62. 6b77990 Fixing build errors on Windows platform. Minor changes... by kjellander@webrtc.org · 13 years ago
  63. fdde8b3 Add references to src/ copies for LICENSE etc. by andrew@webrtc.org · 13 years ago
  64. cb18121 Add an unpacker tool for audioproc debug files. by andrew@webrtc.org · 13 years ago
  65. fc9bcef Data alignment fix for SSIM. by frkoenig@google.com · 13 years ago
  66. 78c767f Rewrote codec test to use fake camera. by phoglund@webrtc.org · 13 years ago
  67. d855c1a Reverts r807 and fixes the real issue in the VCM. by stefan@webrtc.org · 13 years ago
  68. bdb55c8 This CL is an attempt to remove a crash we can see when closing down VoiceEgine. by henrika@webrtc.org · 13 years ago
  69. a6c2335 Solves crash in ADM API unit test for Core Audio on Windows by henrika@webrtc.org · 13 years ago
  70. 5423bc2 Adds correct absolute paths to all input files in ADM functional unit tests. by henrika@webrtc.org · 13 years ago
  71. 5b5c31d Update fixed point audio processing output. by andrew@webrtc.org · 13 years ago
  72. ca325ec Corrected a linux build error introduced in issue 246005. by kma@webrtc.org · 13 years ago
  73. f0cd394 Put fwrite calls under corresponding macros since they shouldn't show up in release build. by wjia@webrtc.org · 13 years ago
  74. f31826e adding a wait on the decode thread when no frames are available by mikhal@webrtc.org · 13 years ago
  75. a412924 VP8:Setting number of cores based on image size by mikhal@webrtc.org · 13 years ago
  76. 913644b For commiting changes in CL 277002, due to file structure changes introduced during by kma@webrtc.org · 13 years ago
  77. 0d0037c Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac). by henrike@webrtc.org · 13 years ago
  78. 0a9c318 The fread result is no longer ignored. by phoglund@webrtc.org · 13 years ago
  79. 537096a Remove unnecessary objective-c compiler flags. by andrew@webrtc.org · 13 years ago
  80. c63f788 Added fake camera, rewrote one test to use it. by phoglund@webrtc.org · 13 years ago
  81. bf478fa Ensures that ADM unit tests builds on all platforms. by henrika@webrtc.org · 13 years ago
  82. 58b4f1c Fixes broken build in peerconnection unit tests by mallinath@webrtc.org · 13 years ago
  83. 4e4c939 Upgrade libvpx to e529a825 by mikhal@webrtc.org · 13 years ago
  84. f1a605c Update DEPS to support Mac clang build. by andrew@webrtc.org · 13 years ago
  85. 5eb64f0 Fix BitrateSent() API when having a default RTP module. by stefan@webrtc.org · 13 years ago
  86. 158f496 Fixes a rate control bug in the VP8 wrapper. by stefan@webrtc.org · 13 years ago
  87. aa32319 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. by perkj@webrtc.org · 13 years ago
  88. ca8b3a3 kind() method in track interface is changed to std::string to keep uniformity with other get methods by mallinath@webrtc.org · 13 years ago
  89. 96ba190 ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. by mallinath@webrtc.org · 13 years ago
  90. ead87b5 Fix potential issue where frame buffers might be freed while being decoded. by stefan@webrtc.org · 13 years ago
  91. 2b0f094 Avoid reallocating the decodedImage for every decoded frame. by stefan@webrtc.org · 13 years ago
  92. ee3dfa6 by mikhal@webrtc.org · 13 years ago
  93. 1af915d video_coding: vp8: Updating error propagation threshold by mikhal@webrtc.org · 13 years ago
  94. 11330b0 Added myself to rtp module watch by pwestin@webrtc.org · 13 years ago
  95. d75889e Change of Android makefiles to build latest video coding code. by kma@webrtc.org · 13 years ago
  96. 7cf8937 by henrika@webrtc.org · 13 years ago
  97. cedbb03 [Issue 101] Solves memory leak on Windows by henrika@webrtc.org · 13 years ago
  98. 2ebc9ce Fix broken PeerConnection Dev build. by perkj@webrtc.org · 13 years ago
  99. c4d1983 Changes in rtp_format_vp8_unittest to match the changes in CL 774. by stefan@webrtc.org · 13 years ago
  100. f553ec7 Notifier and RefCount interface and implementation class name changed according to the naming convention. by mallinath@webrtc.org · 13 years ago