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gerrit.openfyde.cn / webrtc.googlesource.com / src / f75f0cf36a417768a80c64e049c6a657a422eb14
  1. f75f0cf Enable GoogleWifiTrace3Mbps simulations. by Stefan Holmer · 10 years ago
  2. 0d26605 VoE: apply new style guide on VoE interfaces and their implementations by Jelena Marusic · 10 years ago
  3. 79c1433 Delete VoiceChannelTransport before deleting Channel in voe_cmd_test by Minyue Li · 10 years ago
  4. 0b15445 VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/ by Jelena Marusic · 10 years ago
  5. e433c0e Restore back verbosity logging for camera captured frame. by Alex Glaznev · 10 years ago
  6. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
  7. cac1b38 Expose RTCConfiguration to java JNI and add an option to disable TCP by Jiayang Liu · 10 years ago
  8. 4eddf18 Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle. by Peter Thatcher · 10 years ago
  9. 8a6680e Remove base/move.h (no one uses it anymore) by Karl Wiberg · 10 years ago
  10. cbf0927 Revert "rtc::Buffer: Remove backwards compatibility band-aids" by Karl Wiberg · 10 years ago
  11. 9e1a6d7 rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  12. ff019b0 Move rtc::AtomicOps to webrtc/base/atomicops.h. by Peter Boström · 10 years ago
  13. f16fcbe Remove ViECapture usage in VideoSendStream. by Peter Boström · 10 years ago
  14. 46bd31b VoE: VoENetwork unit test by Jelena Marusic · 10 years ago
  15. 3cfa756 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz by Bjorn Volcker · 10 years ago
  16. efbde37 Don't use CPU adaptation for screen content in the new API. by Erik Språng · 10 years ago
  17. adf89b7 Added SetBitRate function to VoE API to allow changing the audio bitrate. by Ivo Creusen · 10 years ago
  18. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  19. 10ba3ee Roll chromium_revision a12e1e1..0cb2549 (326495:327252) by Henrik Kjellander · 10 years ago
  20. dea11f9 Add per flow throughput and delay metrics. by Stefan Holmer · 10 years ago
  21. 94cc1fe Remove ViEImageProcess usage in VideoSendStream. by Peter Boström · 10 years ago
  22. c444de6 Make setup_links.py handle non-link directories during cleanup by Henrik Kjellander · 10 years ago
  23. 1ba344a Adds a MediaConstraint for the AudioOption aec_dump by Bjorn Volcker · 10 years ago
  24. 97f13c5 Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0. by Noah Richards · 10 years ago
  25. 86153c2 Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data. by Noah Richards · 10 years ago
  26. 80154f6 Set correct .type directive for asm functions. by Wei Zhong · 10 years ago
  27. faa6d07 Remove a few verbose log messages from webrtcvideoengine2. by Alex Glaznev · 10 years ago
  28. 019087f Add safeguards against signalling peer-reflexive candidates. by Peter Thatcher · 10 years ago
  29. ae33134 Always specify current OS when syncing Chromium. by Henrik Kjellander · 10 years ago
  30. 8786f63 Roll gtest-parallel. by Peter Boström · 10 years ago
  31. 31dc737 Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel. by Stefan Holmer · 10 years ago
  32. 88de479 AudioEncoderIsac: Print error code if CHECK for successful encoding fails by Karl Wiberg · 10 years ago
  33. bcbcd84 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset. by Stefan Holmer · 10 years ago
  34. 9d657cf Fix dangling pointer in screenshare_loopback by Erik Språng · 10 years ago
  35. beb9798 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool by Bjorn Volcker · 10 years ago
  36. ddbddbd Remove ViENetwork usage in VideoSendStream. by Peter Boström · 10 years ago
  37. 038df3c Remove ViEExternalCodec usage in VideoSendStream. by Peter Boström · 10 years ago
  38. 4a9cb6b Prevent zero-timestamps in captured_frame_. by Peter Boström · 10 years ago
  39. 143cec1 Set correct encoder-specific settings for vpx in the new API. by Erik Språng · 10 years ago
  40. e8a197b Enable isac NEON building on Aarch64 by Zhongwei Yao · 10 years ago
  41. d7e5c44 STUN allocation should not be disabled when using shared port and TURN servers are provided. by Jiayang Liu · 10 years ago
  42. 5a92aa8 Add 3-band filter-bank implementation by Alejandro Luebs · 10 years ago
  43. 494f209 Move CriticalSection into rtc_base_approved. by Tommi · 10 years ago
  44. 59d91dc Remove ViERTP_RTCP usage in VideoSendStream. by Peter Boström · 10 years ago
  45. e6cefb6 GYP variables for building expat, icu, libsrtp, usrsctp by Henrik Kjellander · 10 years ago
  46. 61be2a4 Clean up RTCPSender. by Erik Språng · 10 years ago
  47. 3c391cb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. by Åsa Persson · 10 years ago
  48. 52ef9d7 Stop IncomingVideoStream on delete. by Peter Boström · 10 years ago
  49. 23dc68e Add the rtc_build_openmax_dl variable to the GN build. by Andrew MacDonald · 10 years ago
  50. 12e0329 Do not use Magnifier if there are multiple screens since it sometimes crashes. by Jiayang Liu · 10 years ago
  51. 77d444a Handle the case when hoststring is empty. by Tommi · 10 years ago
  52. c4188fd Use IncomingVideoStream in VideoReceiveStream. by Peter Boström · 10 years ago
  53. f955b5d Add h.264 AVC SPS parsing for resolution (re-land) by Henrik Kjellander · 10 years ago
  54. c043afc Cleanup inside IncomingVideoStream. by Peter Boström · 10 years ago
  55. a9ae0df Roll chromium_revision d5098d0..a12e1e1 (326014:326495) by Henrik Kjellander · 10 years ago
  56. a96f02b Make sure histograms in jitter buffer are only updated if running. by Åsa Persson · 10 years ago
  57. affcfb2 Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16 by Bjorn Volcker · 10 years ago
  58. e3827f2 Revert "Add h.264 AVC SPS parsing for resolution." by Noah Richards · 10 years ago
  59. 5ea8eff Add h.264 AVC SPS parsing for resolution. by Noah Richards · 10 years ago
  60. 9728241 Record H264 NALU type in the h264 header. by Noah Richards · 10 years ago
  61. fe7a80c Prevent sender RTCP signals for receive-only channels. by Peter Boström · 10 years ago
  62. 7f287cc rtc::CriticalSection: Add dummy implementation of IsLocked for release builds by Magnus Jedvert · 10 years ago
  63. 24d4485 Enable -Wunused-private-field warning for talk/ by Henrik Kjellander · 10 years ago
  64. d3e8eda (Re-land) AudioEncoderDecoderIsac: Merge the two config structs by Karl Wiberg · 10 years ago
  65. 92f9eac g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]> by Karl Wiberg · 10 years ago
  66. 261f644 Suppressing VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Dr.Memory by Minyue Li · 10 years ago
  67. 6bf1084 rtc::CriticalSection: Add function IsLocked by Magnus Jedvert · 10 years ago
  68. bd67f66 Restore webrtc/base/move.h, because it's used in Windows Chromium builds by Karl Wiberg · 10 years ago
  69. 3525954 Use short include paths for icu headers. by Henrik Kjellander · 10 years ago
  70. 915590e Moved ByteBuffer/BitBuffer into rtc_base_approved. by Noah Richards · 10 years ago
  71. 01aeaee Fix GetSignatureDigestAlgorithm for openssl to prepare for EC key switch. by JiaYang (佳扬) Liu · 10 years ago
  72. a8e285d Remove webrtc/base/move.h, and make types move-only manually by Karl Wiberg · 10 years ago
  73. ee0b00e Prevent recv-stream reconfig on identical codecs. by Peter Boström · 10 years ago
  74. 908e77b Allow Java code to detect if VP8 and H.264 HW decoding is supported. by Alex Glaznev · 10 years ago
  75. b672882 Move cricket::FakeCall and associates to a separate file. by Fredrik Solenberg · 10 years ago
  76. 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
  77. 96d1d89 Do not register bandwidth observer for receive only channels. by Åsa Persson · 10 years ago
  78. 393347f Report receive-side packet loss. by Peter Boström · 10 years ago
  79. 7c027b6 Enable more Clang warnings for talk/ by Henrik Kjellander · 10 years ago
  80. 5a31780 Reformatting RTPtimeshift.cc file. by Ivo Creusen · 10 years ago
  81. ac69016 Improve TCP by adding a real timeout to in flight packets. by Stefan Holmer · 10 years ago
  82. 8e4b9e8 Roll chromium_revision dcb0929..d5098d0 (325030:326014) by Henrik Kjellander · 10 years ago
  83. e555b7b Fix CC flags in GN Windows build. by Henrik Kjellander · 10 years ago
  84. fb49451 Disables mic bump-up level if not built with chromium by Bjorn Volcker · 10 years ago
  85. 8f85dbc Reduce the number of registers used in MIPS optimizations. by Ljubomir Papuga · 10 years ago
  86. bbf7c86 Add a new BitBuffer class to webrtc base. by Noah Richards · 10 years ago
  87. 61b4d51 Dynamic resolution change for VP8 HW encode. by jackychen · 10 years ago
  88. 5464a6e Remove VideoCodingModule::InitializeReceiver. by Peter Boström · 10 years ago
  89. 9dbbcfb Remove VideoCodingModule::InitializeSender. by Peter Boström · 10 years ago
  90. 9570224 Fix broken perf prints. by Stefan Holmer · 10 years ago
  91. 5f92051 Fix bug in TCP implementation (simulations). by Stefan Holmer · 10 years ago
  92. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  93. 6cff9cf Revert "Remove simulcast modules from ViEReceiver." by Peter Boström · 10 years ago
  94. 06b08af VoE: VoEBase unit test by Jelena Marusic · 10 years ago
  95. c4905fb Fix race condition in Android camera JNI code. by Alex Glaznev · 10 years ago
  96. ac7d97f Remove frame copy in RTCOpenGLVideoRenderer. by Zeke Chin · 10 years ago
  97. 011c00f rtc::Buffer: Accept void* in addition to the byte-sized types by Karl Wiberg · 10 years ago
  98. 8c05415 Add extra logging for Android camera JNI layer. by Alex Glaznev · 10 years ago
  99. 9478437 rtc::Buffer improvements by Karl Wiberg · 10 years ago
  100. 9154373 Do not define POSIX. by Thiago Farina · 10 years ago