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gerrit-public.fairphone.software / platform / external / webrtc / ec869bf7814ae42a79519902aefb254e4bd44d0f
  1. ec869bf Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 11 years ago
  2. e398954 Update usrsctp to r8875 by jiayl@webrtc.org · 11 years ago
  3. 32196de Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  4. a15fbfd Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 11 years ago
  5. 9c09e6e Add high perf mode to VP8 by niklas.enbom@webrtc.org · 11 years ago
  6. 26eaf7c Add a check to all.gyp to respect the include_tests variable. by andrew@webrtc.org · 11 years ago
  7. 2eaac18 Makes the sid of a closed DataChannel available to reuse per the spec. by jiayl@webrtc.org · 11 years ago
  8. a685c9d base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 11 years ago
  9. 5654b30 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 11 years ago
  10. d469443 Rolling new version of opus.gyp by tina.legrand@webrtc.org · 11 years ago
  11. ed3e0d8 Increasing tolerances quite a bit to fight flakes. by phoglund@webrtc.org · 11 years ago
  12. ae740dd (Auto)update libjingle 69359922-> 69365993 by buildbot@webrtc.org · 11 years ago
  13. d42da54 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 11 years ago
  14. 851a09e Initial GN work for WebRTC by kjellander@webrtc.org · 11 years ago
  15. 2ca2188 Restore ptypes.txt file by henrik.lundin@webrtc.org · 11 years ago
  16. 6b06142 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 11 years ago
  17. 8f8503d Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. by minyue@webrtc.org · 11 years ago
  18. 44a317a (Auto)update libjingle 69337301-> 69359922 by buildbot@webrtc.org · 11 years ago
  19. 9f36c08 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. by henrike@webrtc.org · 11 years ago
  20. 53f5793 (Auto)update libjingle 69306183-> 69323802 by buildbot@webrtc.org · 11 years ago
  21. 587ef60 Implement RTP extension support in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  22. d054bff (Auto)update libjingle 69292418-> 69293749 by buildbot@webrtc.org · 11 years ago
  23. d980307 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 11 years ago
  24. 88d9fa6 (Auto)update libjingle 69291002-> 69292418 by buildbot@webrtc.org · 11 years ago
  25. 4b12d40 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 11 years ago
  26. 27626a6 (Auto)update libjingle 69278008-> 69291002 by buildbot@webrtc.org · 11 years ago
  27. d6e2213 Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 11 years ago
  28. 1e3c5c2 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 11 years ago
  29. b099a6f Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 11 years ago
  30. 721f970 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 11 years ago
  31. eb16b81 Implements start bitrate for new video API. by mflodman@webrtc.org · 11 years ago
  32. 0a1e7e0 (Auto)update libjingle 69276003-> 69278008 by buildbot@webrtc.org · 11 years ago
  33. 63e4607 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 11 years ago
  34. 249211e Disable flaky test (WebRtcVideoMediaChannelTest.GetStats) on DrMemory Full. by asapersson@webrtc.org · 11 years ago
  35. d159140 (Auto)update libjingle 69260070-> 69276003 by buildbot@webrtc.org · 11 years ago
  36. 2bae321 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 11 years ago
  37. 117afee (Auto)update libjingle 69188577-> 69260070 by buildbot@webrtc.org · 11 years ago
  38. ab23d49 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 11 years ago
  39. c6c1dfd Add extra logging and latency restriction to VP8 HW encoder. by glaznev@webrtc.org · 11 years ago
  40. a6764ab (Auto)update libjingle 69144530-> 69164179 by buildbot@webrtc.org · 11 years ago
  41. af6f02f Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 11 years ago
  42. db56390 (Auto)update libjingle 69143161-> 69144530 by buildbot@webrtc.org · 11 years ago
  43. f99c2f2 Add NACK feedback parameter to WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  44. e322a17 Implement RTX tests+fixes in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  45. 9fbb717 Remove engine_codecs_ cache from unittests. by pbos@webrtc.org · 11 years ago
  46. d54ec12 Fix GYP DEPTH for libjingle isolate files by kjellander@webrtc.org · 11 years ago
  47. a1bfc50 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 11 years ago
  48. c800c1c (Auto)update libjingle 69131548-> 69132244 by buildbot@webrtc.org · 11 years ago
  49. 1c8223c Initial owners file for talk/media/webrtc/. by pbos@webrtc.org · 11 years ago
  50. 7e71b77 (Auto)update libjingle 69102234-> 69116997 by buildbot@webrtc.org · 11 years ago
  51. 8e256ee Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 11 years ago
  52. 1a6c628 Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' by jiayl@webrtc.org · 11 years ago
  53. 3c13ed3 json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. by henrike@webrtc.org · 11 years ago
  54. ddeec04 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck by jiayl@webrtc.org · 11 years ago
  55. 3f3f428 (Auto)update libjingle 69097619-> 69099564 by buildbot@webrtc.org · 11 years ago
  56. 6c6f33b Fix the flaky RTP DataChannel test. by jiayl@webrtc.org · 11 years ago
  57. 18dfa8d (Auto)update libjingle 69069003-> 69082899 by buildbot@webrtc.org · 11 years ago
  58. cb254aa Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 11 years ago
  59. 27f062a Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  60. 4cb0128 Fixed GetStats when local and remote track are using the same ssrc. by xians@webrtc.org · 11 years ago
  61. 7e3d62d Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 11 years ago
  62. b90619c (Auto)update libjingle 69049090-> 69054765 by buildbot@webrtc.org · 11 years ago
  63. c01cc3d Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 11 years ago
  64. 2881ab1 Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 11 years ago
  65. 276637b Disable flaky test on DrMemory Full. by pbos@webrtc.org · 11 years ago
  66. d41eaeb (Auto)update libjingle 69005149-> 69049090 by buildbot@webrtc.org · 11 years ago
  67. 286cd76 Revert 6405 "Update generated asm offsets scripts." by henrike@webrtc.org · 11 years ago
  68. e9e8007 (Auto)update libjingle 68985065-> 69005149 by buildbot@webrtc.org · 11 years ago
  69. 4aeb941 Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  70. 5b111b0 Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 11 years ago
  71. 8454ad1 Reland: Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 11 years ago
  72. ab85187 Remove unused resource by henrik.lundin@webrtc.org · 11 years ago
  73. 9e65a3b Re-land webrtcmediaengine.cc part of r6397. by pbos@webrtc.org · 11 years ago
  74. fbb567d Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 11 years ago
  75. 5d223a7 (Auto)update libjingle 68982444-> 68983526 by buildbot@webrtc.org · 11 years ago
  76. 6604c6d Revert 6397 "(Auto)update libjingle 68949184-> 68982444" by minyue@webrtc.org · 11 years ago
  77. af214d8 (Auto)update libjingle 68949184-> 68982444 by buildbot@webrtc.org · 11 years ago
  78. e08a11c Revert 6395 "Making WebRTC able to play and record audio to file..." by minyue@webrtc.org · 11 years ago
  79. fa042ca Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 11 years ago
  80. c726b1f Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 11 years ago
  81. 18026ab common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 by bjornv@webrtc.org · 11 years ago
  82. 782978c common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix by bjornv@webrtc.org · 11 years ago
  83. 3f83072 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 11 years ago
  84. e61b8e3 Adds end to end DataChannel tests. by jiayl@webrtc.org · 11 years ago
  85. a40210a Add support for NVidia VP8 HW encoder. by glaznev@webrtc.org · 11 years ago
  86. fd59c39 Delete last file in neteq4 folder by henrik.lundin@webrtc.org · 11 years ago
  87. 919914d MIPS optimizations for ISAC (patch #1) by andrew@webrtc.org · 11 years ago
  88. 0d7ab0a Adding the new video folder and pacer to the wathclist. by mflodman@webrtc.org · 11 years ago
  89. 12cd443 Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 11 years ago
  90. 1014101 Revert 6380 "Replace libjingle_root with talk_root variable." by kjellander@webrtc.org · 11 years ago
  91. 3eb2c2f (Auto)update libjingle 68891947-> 68893961 by buildbot@webrtc.org · 11 years ago
  92. 86f613d Move WebRtcVideoEngine2 fakes to unittest header. by pbos@webrtc.org · 11 years ago
  93. 734a532 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 11 years ago
  94. 0238682 Replace libjingle_root with talk_root variable. by kjellander@webrtc.org · 11 years ago
  95. 7b82c18 Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 11 years ago
  96. 6200481 Create a joint encoder/decoder wrapper for iSAC in ACM by henrik.lundin@webrtc.org · 11 years ago
  97. a90abde Add thread annotations to AcmReceiver by henrik.lundin@webrtc.org · 11 years ago
  98. 190a32f Make some methods in Clock class const declared by henrik.lundin@webrtc.org · 11 years ago
  99. 6b6e58d Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 11 years ago
  100. 85d2794 Adds support for the "apt" format parameter and turns on the RTX feature. by stefan@webrtc.org · 11 years ago