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gerrit-public.fairphone.software / platform / external / webrtc / c21f0c04cc350142076fc4dc4723c23f311ebe81
  1. c21f0c0 Remove WEBRTC_ANDROID from hardcoded gtest relative path usage. by noahric · 9 years ago
  2. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  3. 5af9a28 Roll chromium_revision d131cac..a8b75a6 (357393:357542) by Henrik Kjellander · 9 years ago
  4. 4b938e5 Hide ACMCodecDB::database_ behind accessors by kwiberg · 9 years ago
  5. 1fd4a4a Let AudioCodingModule::SendCodec return Maybe<CodecInst> by kwiberg · 9 years ago
  6. 969aeb1 Revert of Exclude offline bots from CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1420283013/ ) by kjellander · 9 years ago
  7. 5ab193f Remove system_wrappers dep from field_trial_default and metrics_default. by sergeyu · 9 years ago
  8. de94d08 Hide ACMCodecDB::codec_settings_ behind an accessor by kwiberg · 9 years ago
  9. 373284d Make SendStatisticsProxy outlive ViEChannel. by Peter Boström · 9 years ago
  10. 1ba936a Revert of Fix for "Android audio playout doesn't support non-call media stream" (patchset #3 id:40001 of https://codereview.webrtc.org/1419693004/ ) by henrika · 9 years ago
  11. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  12. 5eb9d57 Re-enable PCAP reading in neteq_rtpplay by henrik.lundin · 9 years ago
  13. 32a6eae Exclude offline bots from CQ config. by Henrik Kjellander · 9 years ago
  14. f1104f6 Remove TODO referring to issue1981, which I just marked WontFix. by Andrew MacDonald · 9 years ago
  15. 20a3461 Remove deprecated IsUnresolved() method from SocketAddress API. by tfarina · 9 years ago
  16. 22ae293 Roll chromium_revision 78da654..d131cac (357333:357393) by kjellander · 9 years ago
  17. 5a846c0 Make ConnectionType public in order to add java NetworkObserver. by Honghai Zhang · 9 years ago
  18. 678c903 Delete AcmReceiver::SetInitialDelay by henrik.lundin · 9 years ago
  19. ce4aef1 Adding support for simulcast and spatial layers into VideoQualityTest by sprang · 9 years ago
  20. 8cc126f PRESUBMIT: Enable header guard checks for cpplint. by kjellander · 9 years ago
  21. 1d5c9bd Remove unused method AcmReceiver:RedPayloadType by henrik.lundin · 9 years ago
  22. 792982b Suppress data races in AudioDeviceLinuxPulse::Init. by Peter Boström · 9 years ago
  23. cc41924 Roll chromium_revision 40d9ba6..78da654 (357298:357333) by kjellander · 9 years ago
  24. 6f29a69 Suppress data races in sctp_close. by Peter Boström · 9 years ago
  25. 0b8d056 Rename InitCpuFlags suppression. by Peter Boström · 9 years ago
  26. 9bc2667 ACM/NetEq: Restructure how post-decode VAD is enabled by henrik.lundin · 9 years ago
  27. d56d68c system_wrappers: Fix include header guards. by kjellander · 9 years ago
  28. 2b06352 Roll chromium_revision 0bf3ae4..40d9ba6 (357288:357298) by kjellander · 9 years ago
  29. 608213e Add locks and thread annotations for ReceiverReferenceTimeReportEnabled. by stefan · 9 years ago
  30. a8393d9 Roll chromium_revision d0662bc..0bf3ae4 (357257:357288) by kjellander · 9 years ago
  31. 74f0f35 Delete a chain of methods in ViE, VoE and ACM by henrik.lundin · 9 years ago
  32. e502bbe Update webrtc/base/common.h after recent _DEBUG->!NDEBUG change. by Tommi · 9 years ago
  33. 4040d1e Roll chromium_revision ca6592b..d0662bc (357129:357257) by kjellander · 9 years ago
  34. a41ab93 Switch usage of _DEBUG macro to NDEBUG. by tfarina · 9 years ago
  35. 5c3da4b Call MediaCodec.stop() on separate thread. by Alex Glaznev · 9 years ago
  36. 8e1809f Fix TransientSuppression in audioproc_float by aluebs · 9 years ago
  37. 78858d2 Roll chromium_revision 0ebc3da..ca6592b (357077:357129) by kjellander · 9 years ago
  38. 0be3e04 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on Android. by Henrik Kjellander · 9 years ago
  39. 8a4f547 Hang on android when DNS resolution is not done by Guo-wei Shieh · 9 years ago
  40. 534dafc Roll chromium_revision ce45e11..0ebc3da (357029:357077) by kjellander · 9 years ago
  41. 102c6a6 Replace rtc::cricket::Settable with rtc::Maybe by kwiberg · 9 years ago
  42. bdafe31 Add aecdump support to audioproc_f. by andrew · 9 years ago
  43. 1367fbd Roll chromium_revision 657e8d9..ce45e11 (356260:357029) by kjellander · 9 years ago
  44. cb3f9bd Make the nonlinear beamformer steerable by Alejandro Luebs · 9 years ago
  45. 7367463 Utilize bitrate above codec max to protect video. by pbos · 9 years ago
  46. 315dce7 Enable VP9 internal resize by default. by Marco · 9 years ago
  47. bbaf363 Filter overlapping RTP header extensions. by Stefan Holmer · 9 years ago
  48. 4f5db11 Make VCMEncodedFrameCallback const. by Peter Boström · 9 years ago
  49. 075fb4b MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. by asapersson · 9 years ago
  50. 69ccb33 Remove redudant encoder rate calls. by Peter Boström · 9 years ago
  51. 4f4f756 Create isolate files for nonparallel tests. by Peter Boström · 9 years ago
  52. 1295297 Register header extensions in RtpRtcpObserver to avoid log spam. by Stefan Holmer · 9 years ago
  53. ee1879c Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table by kwiberg · 9 years ago
  54. 48ed930 ACM: Move NACK functionality inside NetEq by henrik.lundin · 9 years ago
  55. a35ae7f Fix chromium-style warnings in webrtc/sound/. by tfarina · 9 years ago
  56. 95192fb Create a 'webrtc_nonparallel_tests' target. by Peter Boström · 9 years ago
  57. 6449990 Update scalability structure data according to updates in the RTP payload profile. by asapersson · 9 years ago
  58. 7464089 audio_coding: rename interface -> include by Henrik Kjellander · 9 years ago
  59. be81fa5 Rewrote perform_action_on_all_files to be parallell. by phoglund · 9 years ago
  60. 32df5ef Update reference indices according to updates in the RTP payload profile. by asapersson · 9 years ago
  61. 1a8240c Disable P2PTransport...TestFailoverControlledSide on Memcheck by Henrik Lundin · 9 years ago
  62. b608eb8 pass clangcl compile options to ignore warnings in gflags.cc by Frank Barchard · 9 years ago
  63. e55c42c Remove limitation on the amount of maximum pending HW decoder inputs. by glaznev · 9 years ago
  64. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  65. ebc0b4e Use webrtc/base/logging.h for rtp_rtcp. by Peter Boström · 9 years ago
  66. 605db69 Disable EndToEndTest.AssignsTrans... for memcheck by henrik.lundin · 9 years ago
  67. 6408174 Fix for "Android audio playout doesn't support non-call media stream" by henrika · 9 years ago
  68. 83585c9 VideoCapturerAndroid: More frequent and verbose logging by magjed · 9 years ago
  69. ec9d187 Added override keyword to overridden methods to stop compiler warnings. by rlester · 9 years ago
  70. fce4a94 RentACodec: New class that takes over part of ACMCodecDB's job by kwiberg · 9 years ago
  71. 77d0d6e When all connections timed out on writing, delete them all. BUG=5111 by honghaiz · 9 years ago
  72. f116bd0 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  73. f1dcd46 UBSan: Add blacklist files for WebRTC standalone. by Henrik Kjellander · 9 years ago
  74. 9397d84 Roll chromium_revision 625f6c8..657e8d9 (356202:356260) by kjellander · 9 years ago
  75. 27f6fd3 Remove noparent from talk/OWNERS. by pbos · 9 years ago
  76. 5ddee02 Landmine: clobber to remove out/{Debug,Release}/args.gn by Henrik Kjellander · 9 years ago
  77. 4f847da Use webrtc/base/checks.h in desktop_capture. by pbos · 9 years ago
  78. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  79. 2a0a2a4 Add stats for used video codec type for a sent video stream: by asapersson · 9 years ago
  80. 18ba3e2 Roll chromium_revision faa5502..625f6c8 (356073:356202) by kjellander · 9 years ago
  81. 18a944b Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ ) by deadbeef · 9 years ago
  82. d3b26d9 Adding the ability to change ICE servers through SetConfiguration. by deadbeef · 9 years ago
  83. 2b55867 Exposing DTLS transport state from TransportChannel. by deadbeef · 9 years ago
  84. b0bb77f Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1416773003/ ) by guoweis · 9 years ago
  85. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  86. aed571f Roll chromium_revision 27af50f..faa5502 (356022:356073) by kjellander · 9 years ago
  87. e2a83ee Introduce rtc::ArrayView<T>, which keeps track of an array that it doesn't own by Karl Wiberg · 9 years ago
  88. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  89. 4cba4eb Disable denoising for VP9 by default. by pbos · 9 years ago
  90. 65e7d4c Remove CanCreateAndDestroyManyVideoStreams. by Peter Boström · 9 years ago
  91. c4ef143 Revert "Add GN Build file for rtc_sound target." by Henrik Kjellander · 9 years ago
  92. 717432f Remove network_enabled_crit_ in call.cc. by mflodman · 9 years ago
  93. 09b38f3 Re-enable VP9 resize test. by Marco · 9 years ago
  94. 7ef0553 Fix for Win GN Build. by tfarina · 9 years ago
  95. 2d3747d Fix for Mac GN BUILD. by tfarina · 9 years ago
  96. e9eca8f Removing AudioCoding class, a.k.a the new ACM API by henrik.lundin · 9 years ago
  97. f054819 Add GN Build file for rtc_sound target. by tfarina · 9 years ago
  98. 213b598 Roll chromium_revision c86a4e2..27af50f (356002:356022) by kjellander · 9 years ago
  99. 415d2cd Use webrtc/base/logging.h for video. by Peter Boström · 9 years ago
  100. f9af108 Roll chromium_revision c708f39..c86a4e2 (355993:356002) by kjellander · 9 years ago