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gerrit-public.fairphone.software / platform / external / webrtc / 87c50f02c69441f4d57402b0c22576c7b1fb5407
  1. 87c50f0 Adding author by niklas.enbom@webrtc.org · 13 years ago
  2. 3a9680b Adding author by niklas.enbom@webrtc.org · 13 years ago
  3. 0ba3133 Aligning license file with file header by niklas.enbom@webrtc.org · 13 years ago
  4. 895870b Adding marker bit to RTPanalyze results by henrik.lundin@webrtc.org · 13 years ago
  5. bb8dfbd updating vpm unit_test following r858 by mikhal@webrtc.org · 13 years ago
  6. 7395d3d Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 by turaj@webrtc.org · 13 years ago
  7. 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
  8. 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
  9. 29d75b3 Only allow increasing capture time. by mflodman@webrtc.org · 13 years ago
  10. 18ee6ec Use __inline in NS-fixed. by andrew@webrtc.org · 13 years ago
  11. 3119ecf Fix audioproc build errors on Windows. by andrew@webrtc.org · 13 years ago
  12. c4ab870 video_processing: Adding logic to avoid a memcpy when not required by mikhal@webrtc.org · 13 years ago
  13. 0ab521f Resolving a crash related to strncopy followed by a strcat by punyabrata@webrtc.org · 13 years ago
  14. 36a992b Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. by perkj@webrtc.org · 13 years ago
  15. d683770 Fixing VPMUnitTest compilation error on Windows. by kjellander@webrtc.org · 13 years ago
  16. b37c628 Fixes crash due to r841. Review URL: http://webrtc-codereview.appspot.com/256004 by henrike@webrtc.org · 13 years ago
  17. e9f909b Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. by kma@webrtc.org · 13 years ago
  18. f1a45d7 Add missing <stdlib.h> to data_log test. by andrew@webrtc.org · 13 years ago
  19. 3134aac Use fileutils for the audio file in voe_auto_test. by andrew@webrtc.org · 13 years ago
  20. 2795750 Changed Android makefile to make the lastest video render code run. by kma@webrtc.org · 13 years ago
  21. 8473688 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  22. 8885d22 by henrike@webrtc.org · 13 years ago
  23. 1e10bb3 Remove global std::strings from fileutils. by andrew@webrtc.org · 13 years ago
  24. 2c74bab Remove unneeded assert and tracing. by andrew@webrtc.org · 13 years ago
  25. 299e2c9 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame by amyfong@webrtc.org · 13 years ago
  26. 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
  27. 177bb52 Fixing system_wrappers unittests. by kjellander@webrtc.org · 13 years ago
  28. 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
  29. 731ecba by henrike@webrtc.org · 13 years ago
  30. 1f6d740 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). by braveyao@webrtc.org · 13 years ago
  31. 88e0a34 Remove duplicated code. Review URL: http://webrtc-codereview.appspot.com/251001 by wu@webrtc.org · 13 years ago
  32. f960211 Fixes two jitter buffer bugs related to NACK. by stefan@webrtc.org · 13 years ago
  33. 35a12cd Fix comment. by perkj@webrtc.org · 13 years ago
  34. 8129752 Add refcount and scoped_refptr. by perkj@webrtc.org · 13 years ago
  35. 94cfde7 Removed scoped_refptr from libjingle.gyp by perkj@webrtc.org · 13 years ago
  36. 7e08613 Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. by perkj@webrtc.org · 13 years ago
  37. 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
  38. eb65860 Reverts the workaround in r823 and solves a macro bug. by stefan@webrtc.org · 13 years ago
  39. 8b1f621 Updated gypi for tests to work on osx. by tina.legrand@webrtc.org · 13 years ago
  40. dfbebb9 Add a documented_interfaces watchlist. by andrew@webrtc.org · 13 years ago
  41. ca4666b vie wintest added hybrid protection mode by amyfong@webrtc.org · 13 years ago
  42. 1e7e60b Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was by amyfong@webrtc.org · 13 years ago
  43. 51e1bb4 vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll by amyfong@webrtc.org · 13 years ago
  44. 5200a05 video_coding/jitter_buffer Updating condition on which we return a frame. by mikhal@webrtc.org · 13 years ago
  45. 30f6376 VP8: Updating codec version: VP8 version will now return the libvpx version used. by mikhal@webrtc.org · 13 years ago
  46. 2d28aff Workaround for an issue where frames are grabbed for decoding prematurely. by stefan@webrtc.org · 13 years ago
  47. fbea4e5 Solves two bandwidth estimation issues and measures the sent video bitrate. by stefan@webrtc.org · 13 years ago
  48. 7e4269e Changed VP8 qp min and added noise reduction. by mflodman@webrtc.org · 13 years ago
  49. 8fc663b Don't trigger false ViE SetReceiveCodec warning. by mflodman@webrtc.org · 13 years ago
  50. 6b77990 Fixing build errors on Windows platform. Minor changes... by kjellander@webrtc.org · 13 years ago
  51. fdde8b3 Add references to src/ copies for LICENSE etc. by andrew@webrtc.org · 13 years ago
  52. cb18121 Add an unpacker tool for audioproc debug files. by andrew@webrtc.org · 13 years ago
  53. fc9bcef Data alignment fix for SSIM. by frkoenig@google.com · 13 years ago
  54. 78c767f Rewrote codec test to use fake camera. by phoglund@webrtc.org · 13 years ago
  55. d855c1a Reverts r807 and fixes the real issue in the VCM. by stefan@webrtc.org · 13 years ago
  56. bdb55c8 This CL is an attempt to remove a crash we can see when closing down VoiceEgine. by henrika@webrtc.org · 13 years ago
  57. a6c2335 Solves crash in ADM API unit test for Core Audio on Windows by henrika@webrtc.org · 13 years ago
  58. 5423bc2 Adds correct absolute paths to all input files in ADM functional unit tests. by henrika@webrtc.org · 13 years ago
  59. 5b5c31d Update fixed point audio processing output. by andrew@webrtc.org · 13 years ago
  60. ca325ec Corrected a linux build error introduced in issue 246005. by kma@webrtc.org · 13 years ago
  61. f0cd394 Put fwrite calls under corresponding macros since they shouldn't show up in release build. by wjia@webrtc.org · 13 years ago
  62. f31826e adding a wait on the decode thread when no frames are available by mikhal@webrtc.org · 13 years ago
  63. a412924 VP8:Setting number of cores based on image size by mikhal@webrtc.org · 13 years ago
  64. 913644b For commiting changes in CL 277002, due to file structure changes introduced during by kma@webrtc.org · 13 years ago
  65. 0d0037c Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac). by henrike@webrtc.org · 13 years ago
  66. 0a9c318 The fread result is no longer ignored. by phoglund@webrtc.org · 13 years ago
  67. 537096a Remove unnecessary objective-c compiler flags. by andrew@webrtc.org · 13 years ago
  68. c63f788 Added fake camera, rewrote one test to use it. by phoglund@webrtc.org · 13 years ago
  69. bf478fa Ensures that ADM unit tests builds on all platforms. by henrika@webrtc.org · 13 years ago
  70. 58b4f1c Fixes broken build in peerconnection unit tests by mallinath@webrtc.org · 13 years ago
  71. 4e4c939 Upgrade libvpx to e529a825 by mikhal@webrtc.org · 13 years ago
  72. f1a605c Update DEPS to support Mac clang build. by andrew@webrtc.org · 13 years ago
  73. 5eb64f0 Fix BitrateSent() API when having a default RTP module. by stefan@webrtc.org · 13 years ago
  74. 158f496 Fixes a rate control bug in the VP8 wrapper. by stefan@webrtc.org · 13 years ago
  75. aa32319 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. by perkj@webrtc.org · 13 years ago
  76. 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
  77. 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
  78. ead87b5 Fix potential issue where frame buffers might be freed while being decoded. by stefan@webrtc.org · 13 years ago
  79. 2b0f094 Avoid reallocating the decodedImage for every decoded frame. by stefan@webrtc.org · 13 years ago
  80. ee3dfa6 by mikhal@webrtc.org · 13 years ago
  81. 1af915d video_coding: vp8: Updating error propagation threshold by mikhal@webrtc.org · 13 years ago
  82. 11330b0 Added myself to rtp module watch by pwestin@webrtc.org · 13 years ago
  83. d75889e Change of Android makefiles to build latest video coding code. by kma@webrtc.org · 13 years ago
  84. 7cf8937 by henrika@webrtc.org · 13 years ago
  85. cedbb03 [Issue 101] Solves memory leak on Windows by henrika@webrtc.org · 13 years ago
  86. 2ebc9ce Fix broken PeerConnection Dev build. by perkj@webrtc.org · 13 years ago
  87. c4d1983 Changes in rtp_format_vp8_unittest to match the changes in CL 774. by stefan@webrtc.org · 13 years ago
  88. f553ec7 Notifier and RefCount interface and implementation class name changed according to the naming convention. by mallinath@webrtc.org · 13 years ago
  89. ae499a2 Set correct codec info before sending frame to VCM. by mflodman@webrtc.org · 13 years ago
  90. 81f25f9 Fixing build errors on Windows platform. Minor changes... by kjellander@webrtc.org · 13 years ago
  91. f3f2f6a * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. by wu@webrtc.org · 13 years ago
  92. 509c9c5 operator + is evaluated before ?: by henrike@webrtc.org · 13 years ago
  93. 4df8c9a by henrike@webrtc.org · 13 years ago
  94. 7ecdf58 Enable chromium_code:1 in the Chrome build. by andrew@webrtc.org · 13 years ago
  95. ffd28f9 Request key frames to battle error propagation. by stefan@webrtc.org · 13 years ago
  96. d0752c3 video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. by mikhal@webrtc.org · 13 years ago
  97. c693bac Only start ViEPerformanceMonitor when needed. by mflodman@webrtc.org · 13 years ago
  98. b5475d0 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. by phoglund@webrtc.org · 13 years ago
  99. 4c63676 Updated the AEC delay logging to output values in ms. PB output updated. by bjornv@webrtc.org · 13 years ago
  100. cc412c1 Remove second instance of ViE PerformanceMonitor. by mflodman@webrtc.org · 13 years ago