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gerrit-public.fairphone.software / platform / external / webrtc / e6bf587259da23e96a8de0957b172fd74c36c3c6
  1. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 10 years ago
  2. db8cf50 Fix two problems in network.cc: by honghaiz · 10 years ago
  3. 1227e8b [rtp_rtcp] time helper functions by danilchap · 10 years ago
  4. 5908c71 Lint fix for webrtc/modules/video_coding PART 3! by philipel · 10 years ago
  5. f5b1abf Roll chromium_revision c844be9..4688e75 (366322:366364) by kjellander · 10 years ago
  6. de94c13 Add webrtc/audio and webrtc/call to WATCHLISTS. by Peter Boström · 10 years ago
  7. 9d3ab61 Lint fix for webrtc/modules/video_coding PART 2! by philipel · 10 years ago
  8. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 10 years ago
  9. cce46fc Lint fix for webrtc/modules/video_coding PART 1! by philipel · 10 years ago
  10. 5380532 Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates. by asapersson · 10 years ago
  11. 9fca7e1 A unittest that reports the statistics for the duration of an APM stream processing API call. by peah · 10 years ago
  12. c693820 CQ: Add linux_libfuzzer_rel trybot as default. by kjellander · 10 years ago
  13. 54bab12 Roll chromium_revision db567a8..c844be9 (366304:366322) by kjellander · 10 years ago
  14. 2f042f2 Roll chromium_revision 1b6c421..db567a8 (365999:366304) by kjellander · 10 years ago
  15. a4df27b Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) by ivoc · 10 years ago
  16. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 10 years ago
  17. bd7d8f7 Adding a MediaStream parameter to createSender. by deadbeef · 10 years ago
  18. 92594a3 Moving FFT on farend signal to where it is used in AEC (bit exact). by minyue · 10 years ago
  19. 4ff818e Make download_from_google_storage print less during runhooks. by kjellander · 10 years ago
  20. 740c367 iSAC: Remove unnecessary WEBRTC_LINUX define. by kjellander · 10 years ago
  21. c155b16 remove deprecated StringToIP() methods from SocketAddress API by tfarina · 10 years ago
  22. 36d4c54 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) by ivoc · 10 years ago
  23. 455a252 Fix pointer compare-and-swap on Windows. by Peter Boström · 10 years ago
  24. c1cd566 delete basictypes.h header by tfarina · 10 years ago
  25. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 10 years ago
  26. 6c6510a audio_device: Move sources into platform-conditions. by kjellander · 10 years ago
  27. 9b7fc7f Defines for ARM and MIPS CPU types. by kjellander · 10 years ago
  28. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 10 years ago
  29. 095ae15 Keep listening if "accept" returns an invalid socket. by jbauch · 10 years ago
  30. 88518a2 Use NV21 instead of YUV12 and clean up. by perkj · 10 years ago
  31. 48477c1 MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture. by perkj · 10 years ago
  32. fc96bd1 Roll chromium_revision e78bc2f..1b6c421 (365856:365999) by kjellander · 10 years ago
  33. 77fa59d Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003 by guoweis · 10 years ago
  34. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 10 years ago
  35. efb047d Compilation failed with openssl. by guoweis · 10 years ago
  36. 933f3ec Roll chromium_revision ddfc1fe..e78bc2f (365801:365856) by kjellander · 10 years ago
  37. 002f0d0 VP9: Set speed setting to 8 for ARM. by Marco · 10 years ago
  38. 5a4ce2f Deleted declaration of VideoCaptureInput::DeliverI420Frame by nisse · 10 years ago
  39. a0b9549 Roll gtest-parallel. by pbos · 10 years ago
  40. 369f828 Adding trace events for the APM render and capture stream processing functions. by peah · 10 years ago
  41. 9390f84 Use std::nullptr_t instead of decltype(nullptr) by kwiberg · 10 years ago
  42. 1e0cfd9 Add VP8 and H264 depacketizer fuzzers. by Peter Boström · 10 years ago
  43. 9d98f21 Roll chromium_revision 68898fb..ddfc1fe (365698:365801) by kjellander · 10 years ago
  44. a689b44 Add tracing to NetEqImpl::InsertPacket by henrik.lundin · 10 years ago
  45. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 10 years ago
  46. e376f0f Add Windows Clang trybots to the default set. by kjellander · 10 years ago
  47. e40dedb Roll chromium_revision 004c7b4..68898fb (365580:365698) by kjellander · 10 years ago
  48. a089257 Cleanup use of "do { ... } while (0)". by torbjorng · 10 years ago
  49. a54a080 Add ufrag to the ICE candidate signaling. by honghaiz · 10 years ago
  50. 3514cbe Add DrFuzz support to webrtc fuzzers. by pbos · 10 years ago
  51. 7cae30c Disable warnings failing when using Clang on Windows. by kjellander · 10 years ago
  52. 9f58795 Roll chromium_revision 2c8eb1f..004c7b4 (365513:365580) by kjellander · 10 years ago
  53. 361888c OWNERS: Add * to .gyp{i,} everywhere. by kjellander@webrtc.org · 10 years ago
  54. 2f29d70 Roll gtest-parallel. by pbos · 10 years ago
  55. 0bc176b Further refactored the echo suppressor code: by peah · 10 years ago
  56. c482eb3 Don't account for audio in the pacer budget. by Stefan Holmer · 10 years ago
  57. 5f026d0 Update NetEq network statistics in neteq_unittest. by minyue · 10 years ago
  58. 4430763 AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders by kwiberg · 10 years ago
  59. 99b1a32 Retyped the frequency estimate of the comfort noise for the higher band to harmonize the AEC code. by peah · 10 years ago
  60. 426ae9d Roll chromium_revision 6e5b8cb..2c8eb1f (365419:365513) by kjellander · 10 years ago
  61. a6db495 Move Rent-A-Codec out of CodecManager by kwiberg · 10 years ago
  62. a29386c Make VoiceDetection not a ProcessingComponent (bit exact). by solenberg · 10 years ago
  63. 672aba3 Fix error prone code in VideoCapturerAndroid by perkj · 10 years ago
  64. 66085be Bugfix that fixes the error where the audio processing module is called by peah · 10 years ago
  65. 54999d4 rtcp::Dlrr block moved into own file and got Parse function by danilchap · 10 years ago
  66. 29e2f93 Fix NoiseSuppression initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1507683006/. by solenberg · 10 years ago
  67. 45fd9fe New macro: RTC_DEPRECATED (for annotating deprecated functions) by kwiberg · 10 years ago
  68. ed644d8 Roll chromium_revision bff4606..6e5b8cb (365226:365419) by kjellander · 10 years ago
  69. eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 10 years ago
  70. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 10 years ago
  71. c1316a1 Fix HPF initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1490333004/. by solenberg · 10 years ago
  72. 95d9851 Add speech encoder to the encoder stack specification struct by kwiberg · 10 years ago
  73. 7eb914d Fix incorrect comment by kwiberg · 10 years ago
  74. 78315b9 Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ ) by Peter Boström · 10 years ago
  75. f9945b2 Only try to pair protocol matching candidates for creating connections. by Honghai Zhang · 10 years ago
  76. 949028f Make LevelEstimation not a ProcessingComponent. by solenberg · 10 years ago
  77. 5e0218c Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ ) by tommi · 10 years ago
  78. 5125433 Android: Refactor renderers to allow apps to inject custom shaders by Magnus Jedvert · 10 years ago
  79. 91941ae rtcp::VoipMetric block moved into own file and got Parse function by danilchap · 10 years ago
  80. 32d989b Disable transport sequence numbers for audio. by Stefan Holmer · 10 years ago
  81. 10aea22 Roll chromium_revision 53970fd..bff4606 (365141:365226) by kjellander · 10 years ago
  82. 377b5e6 enabled cpplint for the webrtc/modules/rtp_rtcp directory by danilchap · 10 years ago
  83. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 10 years ago
  84. 6db6cdc [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs by danilchap · 10 years ago
  85. 9638143 Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ ) by perkj · 10 years ago
  86. e005cf2 [rtp_rtcp] SSRCDatabase class cleaned (including all lint errors) by danilchap · 10 years ago
  87. 5ea3da2 Base webrtc fuzzers on a template. by Peter Boström · 10 years ago
  88. 8f09f17 Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script. by terelius · 10 years ago
  89. 498ae00 Disable ThreadTest.ThreeThreadsInvoke on DrMemory bots. by Stefan Holmer · 10 years ago
  90. 47a740b [rtp_rtcp] lint errors about rand() usage fixed. by danilchap · 10 years ago
  91. 2d36b92 Roll chromium_revision 10bf0e1..53970fd (365000:365141) by kjellander · 10 years ago
  92. 1588793 Fixing flaky LocalP2PTestSctpDataChannel test. by deadbeef · 10 years ago
  93. c9be007 Fixing and re-enabling some flaky PeerConnection tests. by deadbeef · 10 years ago
  94. bd29246 Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ ) by deadbeef · 10 years ago
  95. 82ccfcf Remove unused and rarely used LOG_ macros. by solenberg · 10 years ago
  96. e22e1cb Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ ) by perkj · 10 years ago
  97. 40f349f [rtp_rtcp] Lint errors cleared from rtp_rtcp/test by danilchap · 10 years ago
  98. 3207916 Made EglBase an abstract class and cleaned up. by perkj · 10 years ago
  99. 03960d9 Roll chromium_revision 4bc4277..10bf0e1 (364953:365000) by kjellander · 10 years ago
  100. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 10 years ago