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webrtc / src / 143cec1cc68b9ba44f3ef4467f1422704f2395f0
  1. 143cec1 Set correct encoder-specific settings for vpx in the new API. by Erik Språng · 10 years ago
  2. e8a197b Enable isac NEON building on Aarch64 by Zhongwei Yao · 10 years ago
  3. d7e5c44 STUN allocation should not be disabled when using shared port and TURN servers are provided. by Jiayang Liu · 10 years ago
  4. 5a92aa8 Add 3-band filter-bank implementation by Alejandro Luebs · 10 years ago
  5. 494f209 Move CriticalSection into rtc_base_approved. by Tommi · 10 years ago
  6. 59d91dc Remove ViERTP_RTCP usage in VideoSendStream. by Peter Boström · 10 years ago
  7. e6cefb6 GYP variables for building expat, icu, libsrtp, usrsctp by Henrik Kjellander · 10 years ago
  8. 61be2a4 Clean up RTCPSender. by Erik Språng · 10 years ago
  9. 3c391cb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. by Åsa Persson · 10 years ago
  10. 52ef9d7 Stop IncomingVideoStream on delete. by Peter Boström · 10 years ago
  11. 23dc68e Add the rtc_build_openmax_dl variable to the GN build. by Andrew MacDonald · 10 years ago
  12. 12e0329 Do not use Magnifier if there are multiple screens since it sometimes crashes. by Jiayang Liu · 10 years ago
  13. 77d444a Handle the case when hoststring is empty. by Tommi · 10 years ago
  14. c4188fd Use IncomingVideoStream in VideoReceiveStream. by Peter Boström · 10 years ago
  15. f955b5d Add h.264 AVC SPS parsing for resolution (re-land) by Henrik Kjellander · 10 years ago
  16. c043afc Cleanup inside IncomingVideoStream. by Peter Boström · 10 years ago
  17. a9ae0df Roll chromium_revision d5098d0..a12e1e1 (326014:326495) by Henrik Kjellander · 10 years ago
  18. a96f02b Make sure histograms in jitter buffer are only updated if running. by Åsa Persson · 10 years ago
  19. affcfb2 Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16 by Bjorn Volcker · 10 years ago
  20. e3827f2 Revert "Add h.264 AVC SPS parsing for resolution." by Noah Richards · 10 years ago
  21. 5ea8eff Add h.264 AVC SPS parsing for resolution. by Noah Richards · 10 years ago
  22. 9728241 Record H264 NALU type in the h264 header. by Noah Richards · 10 years ago
  23. fe7a80c Prevent sender RTCP signals for receive-only channels. by Peter Boström · 10 years ago
  24. 7f287cc rtc::CriticalSection: Add dummy implementation of IsLocked for release builds by Magnus Jedvert · 10 years ago
  25. 24d4485 Enable -Wunused-private-field warning for talk/ by Henrik Kjellander · 10 years ago
  26. d3e8eda (Re-land) AudioEncoderDecoderIsac: Merge the two config structs by Karl Wiberg · 10 years ago
  27. 92f9eac g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]> by Karl Wiberg · 10 years ago
  28. 261f644 Suppressing VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Dr.Memory by Minyue Li · 10 years ago
  29. 6bf1084 rtc::CriticalSection: Add function IsLocked by Magnus Jedvert · 10 years ago
  30. bd67f66 Restore webrtc/base/move.h, because it's used in Windows Chromium builds by Karl Wiberg · 10 years ago
  31. 3525954 Use short include paths for icu headers. by Henrik Kjellander · 10 years ago
  32. 915590e Moved ByteBuffer/BitBuffer into rtc_base_approved. by Noah Richards · 10 years ago
  33. 01aeaee Fix GetSignatureDigestAlgorithm for openssl to prepare for EC key switch. by JiaYang (佳扬) Liu · 10 years ago
  34. a8e285d Remove webrtc/base/move.h, and make types move-only manually by Karl Wiberg · 10 years ago
  35. ee0b00e Prevent recv-stream reconfig on identical codecs. by Peter Boström · 10 years ago
  36. 908e77b Allow Java code to detect if VP8 and H.264 HW decoding is supported. by Alex Glaznev · 10 years ago
  37. b672882 Move cricket::FakeCall and associates to a separate file. by Fredrik Solenberg · 10 years ago
  38. 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
  39. 96d1d89 Do not register bandwidth observer for receive only channels. by Åsa Persson · 10 years ago
  40. 393347f Report receive-side packet loss. by Peter Boström · 10 years ago
  41. 7c027b6 Enable more Clang warnings for talk/ by Henrik Kjellander · 10 years ago
  42. 5a31780 Reformatting RTPtimeshift.cc file. by Ivo Creusen · 10 years ago
  43. ac69016 Improve TCP by adding a real timeout to in flight packets. by Stefan Holmer · 10 years ago
  44. 8e4b9e8 Roll chromium_revision dcb0929..d5098d0 (325030:326014) by Henrik Kjellander · 10 years ago
  45. e555b7b Fix CC flags in GN Windows build. by Henrik Kjellander · 10 years ago
  46. fb49451 Disables mic bump-up level if not built with chromium by Bjorn Volcker · 10 years ago
  47. 8f85dbc Reduce the number of registers used in MIPS optimizations. by Ljubomir Papuga · 10 years ago
  48. bbf7c86 Add a new BitBuffer class to webrtc base. by Noah Richards · 10 years ago
  49. 61b4d51 Dynamic resolution change for VP8 HW encode. by jackychen · 10 years ago
  50. 5464a6e Remove VideoCodingModule::InitializeReceiver. by Peter Boström · 10 years ago
  51. 9dbbcfb Remove VideoCodingModule::InitializeSender. by Peter Boström · 10 years ago
  52. 9570224 Fix broken perf prints. by Stefan Holmer · 10 years ago
  53. 5f92051 Fix bug in TCP implementation (simulations). by Stefan Holmer · 10 years ago
  54. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  55. 6cff9cf Revert "Remove simulcast modules from ViEReceiver." by Peter Boström · 10 years ago
  56. 06b08af VoE: VoEBase unit test by Jelena Marusic · 10 years ago
  57. c4905fb Fix race condition in Android camera JNI code. by Alex Glaznev · 10 years ago
  58. ac7d97f Remove frame copy in RTCOpenGLVideoRenderer. by Zeke Chin · 10 years ago
  59. 011c00f rtc::Buffer: Accept void* in addition to the byte-sized types by Karl Wiberg · 10 years ago
  60. 8c05415 Add extra logging for Android camera JNI layer. by Alex Glaznev · 10 years ago
  61. 9478437 rtc::Buffer improvements by Karl Wiberg · 10 years ago
  62. 9154373 Do not define POSIX. by Thiago Farina · 10 years ago
  63. d43ba89 PRESUBMIT: Add new trybots for iOS Simulator by Henrik Kjellander · 10 years ago
  64. 599beb8 Revert "AudioEncoderDecoderIsac: Merge the two config structs" by Ted Nakamura · 10 years ago
  65. 09a9ea8 Supporting formats of non-multiple of 16 widths on Android. by Henrik Boström · 10 years ago
  66. a51e8f4 Fix some simulation issues. by Stefan Holmer · 10 years ago
  67. 14a97f0 Remove simulcast modules from ViEReceiver. by Peter Boström · 10 years ago
  68. 1d19893 Add TCP fairness test. by Stefan Holmer · 10 years ago
  69. b0b5425 Let rtp_analyze parse absolute sender time by Henrik Lundin · 10 years ago
  70. 61c2a6f Remove rtc::Buffer::length(), since no one uses it anymore by Karl Wiberg · 10 years ago
  71. d4e8014 Fix build errors in r9022 / 09bdc1e5f5a9. by Stefan Holmer · 10 years ago
  72. 09bdc1e Add a BWE fairness test. by Stefan Holmer · 10 years ago
  73. 3795937 Adds a simplified Reno-type TCP sender. by Stefan Holmer · 10 years ago
  74. f49dbfa Close all camera resources when camera error happens. by Alex Glaznev · 10 years ago
  75. 3f4eed0 Deliver RTCP packets only once per receive stream. by Peter Boström · 10 years ago
  76. fb98c40 Register RTP/RTCP modules outside rtp_rtcp_cs_. by Peter Boström · 10 years ago
  77. 382c58d Move target_subarch from gyp_webrtc to supplement.gypi by Henrik Kjellander · 10 years ago
  78. f2497cf Fix unknown option '-msse2' warning by Henrik Kjellander · 10 years ago
  79. 7c324ca AudioEncoderDecoderIsac: Merge the two config structs by Karl Wiberg · 10 years ago
  80. 9829af4 Disable VP8 encoder HW acceleration on NVidia devices. by Alex Glaznev · 10 years ago
  81. 7d89f80 Use BoringSSL as default on iOS by Zeke Chin · 10 years ago
  82. 5d22c00 Add performance tests flag to audioproc_float by Alejandro Luebs · 10 years ago
  83. 41ee1ea Modified the simulcast encoder adapter to correctly handle encoded frames from sub encoders even if the encoder is unable to (temporarily or permanently) produce frames of the exactly matching resolution. This is done by using a different EncodedImageCallback for each encoder, which remembers which VideoEncoder it is registered to and forwards that on to SimulcastEncoderAdapter::Encoded. by Noah Richards · 10 years ago
  84. 099323e Have ViE sender also use the last encoded frame timestamp when determining if the video stream is paused/muted, for purposes of padding. by Noah Richards · 10 years ago
  85. 352b2d7 Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream). by Åsa Persson · 10 years ago
  86. 4b76c02 Roll chromium_revision 8af41b3..dcb0929 (324854:325030) by Magnus Jedvert · 10 years ago
  87. 3c3f646 Prevent null-stream reconfigs on RTP extensions. by Peter Boström · 10 years ago
  88. 36fc1ba Update renamed Android ARM64 trybot in PRESUBMIT.py. by Henrik Kjellander · 10 years ago
  89. c317ce5 VoE: move mock directory 1 level up by Jelena Marusic · 10 years ago
  90. adc46c4 audio_processing/agc: Adds config to set minimum microphone volume at startup by Bjorn Volcker · 10 years ago
  91. 19a3807 Updated .gitignore to ignore isolate_deps_dir. by Henrik Boström · 10 years ago
  92. a9c0ae2 Add a sparse FIR filter implementation by Alejandro Luebs · 10 years ago
  93. e432800 Enable CPU adaptation by default. by Peter Boström · 10 years ago
  94. fcf54bd Reland "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
  95. 73ba7a6 Remove PORTALLOCATOR_ENABLE_BUNDLE, PortAllocatorSessionProxy, PortAllocatorSessionMuxer, and PortProxy. by Peter Thatcher · 10 years ago
  96. 74b9769 Deliver RTCP packets only once per send stream. by Peter Boström · 10 years ago
  97. 2dd6a27 VoE: format VoEBase according to new style guide by Jelena Marusic · 10 years ago
  98. 0de7bcf Removes use of AudioManager.setSpeakerphoneOn in audio manager by henrika · 10 years ago
  99. 6739952 Roll chromium_revision 70a0480..ac81bcc (324430:324836) by Henrik Kjellander · 10 years ago
  100. 56d5028 Remove SignalCaptureStateChange from MediaEngine. by Peter Thatcher · 10 years ago