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webrtc / src / / 6979b024d7cebfdcd1e8f66da59ea157bbc9e47e
  1. 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 10 years ago
  2. 4ba059d Remove custom handler since the logger already logs to console by default. by Jiayang Liu · 10 years ago
  3. 8937437 Do not prune if the current best connection is weak. by honghaiz · 10 years ago
  4. ea70d77 VideoCapturerAndroid: Add test for making calls on stopped camera by Magnus Jedvert · 10 years ago
  5. 59e72ab Enable logging for Mac by default on debug builds. by deadbeef · 10 years ago
  6. f4d38ea CodecOwner: Don't look at definitions for classes we don't link with by kwiberg · 10 years ago
  7. 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 10 years ago
  8. 5e9a1bc Revert of Android GlRectDrawer: Add test for RGB rendering (patchset #3 id:40001 of https://codereview.webrtc.org/1367923002/ ) by magjed · 10 years ago
  9. a58ea78 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any. by honghaiz · 10 years ago
  10. 8a88dd2 Stability improvement for audio recording on Android by henrika · 10 years ago
  11. 2bc68c7 Wire up QualityScaler for H.264 on Android. by Peter Boström · 10 years ago
  12. 7076729 Enable SurfaceViewRenderer for AppRTCDemo by Magnus Jedvert · 10 years ago
  13. 9236bb1 Minor fix for improving logging of supported platform effects by henrika · 10 years ago
  14. 6b8d355 Reland "Wire up send-side bandwidth estimation." by Erik Språng · 10 years ago
  15. 8c266e6 H264 bitstream parser. by Peter Boström · 10 years ago
  16. 6b20ad9 Android GlRectDrawer: Add test for RGB rendering by Magnus Jedvert · 10 years ago
  17. 2efe58b VideoCapturerAndroidTest: Dispose PeerConnectionFactory with pending frames by Magnus Jedvert · 10 years ago
  18. ec249d4 ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace by kwiberg · 10 years ago
  19. 4a3ccad Remove SetAudioDelayOffset() and friends. by solenberg · 10 years ago
  20. f66a925 Don't link with audio codecs that we don't use by kwiberg · 10 years ago
  21. 61e933e Remove ChannelManager::GetCapabilities() by solenberg · 10 years ago
  22. c675ddd video_capture: Better support for UYVY by will.newton · 10 years ago
  23. 74d85e1 Reduce locking in overuse frame detector now that (as of r9508) the observer_ and options_ can only be set at construction time. E.g. no lock is any longer held while doing the callback. by asapersson · 10 years ago
  24. facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 10 years ago
  25. 7603c76 Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ ) by deadbeef · 10 years ago
  26. 70702af Adding PeerConnectionInterface::SetConfiguration method. by deadbeef · 10 years ago
  27. 53eee43 Address the comment from 1367553002. by Guo-wei Shieh · 10 years ago
  28. 2e4b620 TcpPort doesn't connect when calling gmail with non-proxied UDP disabled. by Guo-wei Shieh · 10 years ago
  29. cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 10 years ago
  30. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 10 years ago
  31. d0b5b09 Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc. by Fredrik Solenberg · 10 years ago
  32. 7cf0445 Remove ViEChannel::StartSend deadlock suppression. by Peter Boström · 10 years ago
  33. b7af7b0 Add myself to watchlist for a few subtrees of the repo. by Fredrik Solenberg · 10 years ago
  34. 8bffba7 Fix BWE bug where audio has timestamps in us. by Stefan Holmer · 10 years ago
  35. 6d92bf5 Returning correct duration estimate on Opus DTX packets. by minyuel · 10 years ago
  36. c14f5ff Improving support for Android Audio Effects in WebRTC. by henrika · 10 years ago
  37. c9bbeb0 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ ) by Erik Språng · 10 years ago
  38. d5c75b1 Reduce LS_INFO spam from voice_engine/. by Peter Boström · 10 years ago
  39. 7d17336 Remove the [Un]RegisterVoiceProcessor() API. by Fredrik Solenberg · 10 years ago
  40. 0967734 Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used. by Fredrik Solenberg · 10 years ago
  41. f706c8a VideoCapturerAndroid: Fix threading issues by Magnus Jedvert · 10 years ago
  42. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 10 years ago
  43. 2d4e6c5 Fixing camera capture for video_loopback by ivica · 10 years ago
  44. 47ee2f3 TransportController refactoring. by deadbeef · 10 years ago
  45. 8967183 Simple cleanups of AudioDecoder and AudioEncoder classes by kwiberg · 10 years ago
  46. c1a1b35 Remove the SetLocalMonitor() API. by solenberg · 10 years ago
  47. 07d0936 Purge nss files and dependencies. by torbjorng · 10 years ago
  48. 7404368 Move AudioDecoderIsac* to its own files by Karl Wiberg · 10 years ago
  49. 7083e11 Remove callback_cs_ in ViEEncoder. by Peter Boström · 10 years ago
  50. 8212265 Android: Add class ThreadUtils with helper function joinUninterruptibly() by Magnus Jedvert · 10 years ago
  51. 6faf5be Move AudioDecoderPcm* next to AudioEncoderPcm* by kwiberg · 10 years ago
  52. d4818e7 Using static frame generator when no scrolling by ivica · 10 years ago
  53. 9b5476d sslidentity.cc/IntKeyTypeFamilyToKeyType function added, converting from int to KeyType. by Henrik Boström · 10 years ago
  54. ef165eef Wire up send-side bandwidth estimation. by sprang · 10 years ago
  55. 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 10 years ago
  56. 7317248 Rename CaptureThread to EncodingThread. by Peter Boström · 10 years ago
  57. ef5d5e4 Add field trial for automic resize in MediaCodecVideoEncoder. by asapersson · 10 years ago
  58. 1356ba5 Fixing target_bitrate_bps for a FullStackTest by ivica · 10 years ago
  59. 8f4f00f CQ: Update trybots by Henrik Kjellander · 10 years ago
  60. e4ba6ce9 Log the tag in native log stream. by Jiayang Liu · 10 years ago
  61. ebbf8a8 Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. by sprang · 10 years ago
  62. 04ac81f Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). by Peter Thatcher · 10 years ago
  63. 5bfc6cb Revert "Android: Enable C99 mode instead of C89 (default)." by Henrik Kjellander · 10 years ago
  64. 275a2f1 Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ ) by tommi · 10 years ago
  65. ae16f85 Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). by honghaiz · 10 years ago
  66. c19922c Android SurfaceViewRenderer: Block in release() until frames are returned and cleanup is done by Magnus Jedvert · 10 years ago
  67. e6d3ada Re-add SurfaceTexture as member for setLocalPreview in VideoCapturerAndroid. by Per · 10 years ago
  68. 40bf493 Revert of Update build files to use webrtc_overrides in Chromium instead of overrides. (patchset #2 id:20001 of https://codereview.webrtc.org/1354933002/ ) by henrikg · 10 years ago
  69. 780be75 Make PeerConnectionTest.doTest wait for ice candidates by perkj · 10 years ago
  70. baae0a8 Update build files to use webrtc_overrides in Chromium instead of overrides. by henrikg · 10 years ago
  71. 35d1767 Remove the video capture module on Android. by perkj · 10 years ago
  72. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 10 years ago
  73. 9af63f4 TransportController refactoring. by deadbeef · 10 years ago
  74. 2803a40 Fix ChromeOS build (C99 break) by henrikg · 10 years ago
  75. 4a78308 Android: Add helper class GlTextureFrameBuffer by Magnus Jedvert · 10 years ago
  76. e1aa5b5 This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e. by Ivo Creusen · 10 years ago
  77. ca14b2f Add system log fallback when native logging is unavailable. by jiayl · 10 years ago
  78. e510d7f Remove ACM AudioCodingFeedback callback object and derived classes by henrik.lundin · 10 years ago
  79. be49595 Revert of Tool to convert RtcEventLog files to RtpDump format. (patchset #11 id:200001 of https://codereview.webrtc.org/1297653002/ ) by henrikg · 10 years ago
  80. f4aa4c2 Remove id from VideoProcessingModule. by Peter Boström · 10 years ago
  81. 3520f9e Removes camera.setPreviewTexture in doStopCaptureOnCameraThread and removes the try catch statement since the only method throwing an exception was setPreviewTexture. by Per · 10 years ago
  82. 586b19b Enable probing with repeated payload packets by default. by Stefan Holmer · 10 years ago
  83. 71df77b Remove overridden basictypes.h. by henrikg · 10 years ago
  84. 061b79a ACM: Remove functions related to DTMF by henrik.lundin · 10 years ago
  85. 11d583f Fix a bug in RtpFileSource related to RTCP packets in rtpdump files by henrik.lundin · 10 years ago
  86. 35624c2 Tool to convert RtcEventLog files to RtpDump format. by Ivo Creusen · 10 years ago
  87. 7cbd188 Remove GICE (again). by Peter Thatcher · 10 years ago
  88. ac547a6 Remove channel ids from various interfaces. by Peter Boström · 10 years ago
  89. 1d5198d Fix parameter in VP9 resize test. by Marco · 10 years ago
  90. f350720 VP9: Add automaticeResize to codec setting. by Marco · 10 years ago
  91. e1c5ec7 Fixing bad merge (CHECK is now RTC_CHECK) by Patrik Höglund · 10 years ago
  92. fdd1b9a Reland: Bailing out if pc factory fails to get created. by Patrik Höglund · 10 years ago
  93. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 10 years ago
  94. ae856f2 Added support for logging the SSRC corresponding to AudioPlayout events. by Ivo Creusen · 10 years ago
  95. 48c46db Reduces default sample rate from 44.1kHz to 16kHz to ensure by henrika · 10 years ago
  96. d2320ce CQ: Remove baremetal machines from CQ bots. by Henrik Kjellander · 10 years ago
  97. 5d6a06c Refactoring full stack and loopback tests by ivica · 10 years ago
  98. f2bfc2b Remove some dead code. by Peter Boström · 10 years ago
  99. e64fbce Changed loopback transport in RtxNackTest to not store sequence numbers for retransmitted packets. by terelius · 10 years ago
  100. ada4c13 Move AudioDecoderG722 next to AudioEncoderG722 by kwiberg · 10 years ago