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webrtc / src.git / b69ab79338bff71ea411b82f3dd59508617a11d5
  1. b69ab79 VideoCapturerAndroid: Add function to change capture format while camera is running by magjed · 10 years ago
  2. 496019c If the array size is even, the median should be average of its two middlemost elements. by Cesar Magalhaes · 10 years ago
  3. 83d6b0c2 Ignore genperf lib in merge_libs.py. by noahric · 10 years ago
  4. 343714e Fix the problom that on Linux no external audio device can be selected since #9243. by Brave Yao · 10 years ago
  5. 2981945 Moved arrray_util include to beamformer.h by bloch · 10 years ago
  6. 8ff04d6 Remove UpdateSsrcs from EncoderStateFeedback. by pbos · 10 years ago
  7. 324d9c9 Avoids error message about unknown selected data source for Port iPhone Microphone by henrika · 10 years ago
  8. f421bdc Fix an NPE when creating TurnPort with a NULL socket. by honghaiz · 10 years ago
  9. be37888 Fixing scenario where track is rejected and later un-rejected. by deadbeef · 10 years ago
  10. b947f28 Add pcap support to bwe tools. Allow filtering on SSRCs. by stefan · 10 years ago
  11. fabe2c9 Remove deprecated functions. by jbauch · 10 years ago
  12. c27d89f Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame. by qiangchen · 10 years ago
  13. c5d0d95 Ensuring that UDP TURN servers are always used as STUN servers. by deadbeef · 10 years ago
  14. d848d5e Enable cropping window capturing for Win7 when Aero is disabled. by Jiayang Liu · 10 years ago
  15. bd38428 Don't use result of "field_trial::FindFullName" as string reference. by jbauch · 10 years ago
  16. a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 10 years ago
  17. 083b73f Use std::string references instead of copying contents. by jbauch · 10 years ago
  18. cd67022 Define Stream base classes by Jelena Marusic · 10 years ago
  19. cddb367 Remove unused metric in overuse detector. by Asa Persson · 10 years ago
  20. f393829 Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used. by deadbeef · 10 years ago
  21. fb19f49 Replaced uint32_t with standard uint16_t for sequence_number variables. by Cesar Magalhaes · 10 years ago
  22. bf40b42 Modified Simulation Framework Jitter Model. by Cesar Magalhaes · 10 years ago
  23. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 10 years ago
  24. 9c261f2 Supports logging for dynamic and histogram plots on Simulation Framework. by Cesar Magalhaes · 10 years ago
  25. a4a8d4a Base padding bitrate for an encoder on the bitrate allocated for that encoder, rather than the total bitrate of the channel group. by stefan · 10 years ago
  26. 3258db2 Split iSAC encoder/decoder: Test more cases (and make sure they work) by kwiberg · 10 years ago
  27. 2d3b7e2 AppRTCDemo file logging. by Zeke Chin · 10 years ago
  28. 43e7d3b Avoid overflow in checking for emulation bytes in rbsp. by noahric · 10 years ago
  29. ba8c15b Merge methods for configuring NACK/FEC/hybrid. by pbos · 10 years ago
  30. caa498a Make sure RTCP is sent in tests when receiving packets even if REMB is delayed. by stefan · 10 years ago
  31. ba35d05 Cleanup of iOS AudioDevice implementation by henrika · 10 years ago
  32. d6f1a38 Remove ViEChannel simulcast lock. by Peter Boström · 10 years ago
  33. 4988ca5 Removed unused variables and the need to include the d3dx9.h file. by dkirovbroadsoft · 10 years ago
  34. 870eee4 Fix simulator issue where chokes didn't apply to non-congested packets. by stefan · 10 years ago
  35. a03cd3f 1. Override and virtual has to be consistent. by honghaiz · 10 years ago
  36. 6e2ce6e Allow for framerate reduction for HW encoder. by jackychen · 10 years ago
  37. 9009962 Add methods to set the ICE connection receiving_timeout values. by honghaiz · 10 years ago
  38. 45d1fde Revert of Fix simulator issue where chokes didn't apply to non-congested packets. (patchset #2 id:20001 of https://codereview.webrtc.org/1233853002/) by stefan · 10 years ago
  39. 662ae00 Fix simulator issue where chokes didn't apply to non-congested packets. by Stefan Holmer · 10 years ago
  40. 5d6e58e Improvements to rtc::Bind by Jelena Marusic · 10 years ago
  41. 30409b4 Add statistics gathering for packet loss. by bcornell · 10 years ago
  42. 35b72fb Add new variance update option and unittests for intelligibility by ekm · 10 years ago
  43. d10a68e Don't create unsignalled receive streams for RTX, RED RTX, and ULPFEC packets. by noahric · 10 years ago
  44. 8647922 Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545. by Stefan Holmer · 10 years ago
  45. a6d2444 Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code. by Peter Thatcher · 10 years ago
  46. 4d9d097 Fix follow-up in webrtc/test/field_trial.cc. by pbos · 10 years ago
  47. 97f44e1 Remove a superfluous qualifier on an inline method. by thakis · 10 years ago
  48. 50cf10d Make .gni less sensitive to type of arm_use_neon flags by petermayo · 10 years ago
  49. 11324b9 Wait for a longer time (5 seconds) before establishing the first bandwidth estimate. by Stefan Holmer · 10 years ago
  50. bb36fdf Remove empty-string comparisons. by pbos · 10 years ago
  51. 3b1e647b6 Remove media sinks from Channel. by pbos · 10 years ago
  52. 0f620f4 Make sure we process all pending offer/answer requests before terminating. by tommi · 10 years ago
  53. 31acf3d Add include_examples GYP variable. by kjellander · 10 years ago
  54. e987a47 Removed some unused variables in Windows code. by mgiuca · 10 years ago
  55. 6109386 Expose the disable encryption option to JNI. by Jiayang Liu · 10 years ago
  56. 5436051 Add flakyness check based on the recently received packets. by Peter Thatcher · 10 years ago
  57. aa97df4 Roll chromium_revision 3ead4bc..f8d6ba9 (336983:337800) by tnakamura · 10 years ago
  58. cbd44e6 Use Resampler default constructor in VAD by aluebs · 10 years ago
  59. b8b0143 Tighten link-local routing exclusion check by Peter Thatcher · 10 years ago
  60. 6e89b25 VP9 wrapper: Adjust speed setting. by Marco · 10 years ago
  61. d436298 Remove ResetStatistics from RTP feedback. by pbos · 10 years ago
  62. 19492f1 Add scoped class for overriding field trials. by pbos · 10 years ago
  63. a7d7054 Remove VCM_*_PAYLOAD_TYPE constants. by pbos · 10 years ago
  64. c62642c Make the BWE threshold adaptive. by stefan · 10 years ago
  65. 4e7aa43 audio_processing: Adds two UMA histograms logging delay jumps in AEC by Bjorn Volcker · 10 years ago
  66. f935bcc Use strcmp instead of == operator for c.name and name to find appropriate classes for WebRtcAudio*.java by sophiechang · 10 years ago
  67. 2bad88d Prevent heap overflows for incorrect FEC packet lengths. by pbos · 10 years ago
  68. 468e62a Remove MimdRateControl and factories for RemoteBitrateEstimor. by Erik Språng · 10 years ago
  69. d92f267 audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms by Bjorn Volcker · 10 years ago
  70. 72a8cee Targets should not depend on protobuf when enable_protobuf=0. by André Susano Pinto · 10 years ago
  71. 894ad94 Fix occurrences of const typed declaration without initialization by eblima · 10 years ago
  72. ac8869e Report metrics about negotiated ciphers. by jbauch · 10 years ago
  73. 366e952 Follow-up: Remove old ReportedDelay AEC config by henrik.lundin · 10 years ago
  74. 2224294 iSAC: Functions for importing and exporting bandwidth est. info by Karl Wiberg · 10 years ago
  75. cd4a9bd Remove decoder-thread instantiation for senders. by pbos · 10 years ago
  76. db0cf76 Add test for dropping repeated NTP timestamps. by pbos · 10 years ago
  77. f4eca64 iSAC: Pad with zeros instead of random data, to make testing easier by kwiberg · 10 years ago
  78. 0f133b9 Rename APM Config ReportedDelay to DelayAgnostic by henrik.lundin · 10 years ago
  79. 0d7dbde Update AppRTCDemo resolution for iPhone6/6+ by tkchin · 10 years ago
  80. a771bf8 Fix some clang warnings with -Wmissing-braces in WebRTC. by dcheng · 10 years ago
  81. d830aea Add tkchin to video_coding OWNERS. by Zeke Chin · 10 years ago
  82. 0edd50c Support for onbufferedamountlow by bemasc · 10 years ago
  83. 545727e Move early-return in TimeToSendPadding. by pbos · 10 years ago
  84. bd2522a Fail RTP parsing on excessive padding length. by pbos · 10 years ago
  85. 8b80fb6 Roll chromium_revision fbf756f..3ead4bc (336289:336983) by phoglund · 10 years ago
  86. 4daa90e Prevent size_t underflow in H264 SPS parsing. by pbos · 10 years ago
  87. 2f15093 Prevent OOB read on truncated H264 headers. by pbos · 10 years ago
  88. 7ada923 Prevent OOB reads for zero-length H264 payloads. by pbos · 10 years ago
  89. 48c3839 Prevent depacketizer OOB reads on zero-length VP8 payload. by pbos · 10 years ago
  90. 6e355af Added fields for configuration information to the protobuf format by terelius · 10 years ago
  91. 2e43b26 Prevent OOB reads in FEC packets without complete RED headers. by pbos · 10 years ago
  92. 1adbacb Adding method IsInBeam to beamformer class. by bloch · 10 years ago
  93. 3c60d61 Remove a cast again, after it was shown to worsen Windows perf. by pkasting · 10 years ago
  94. 71f6f44 iOS HW H264 support. by Zeke Chin · 10 years ago
  95. 70d5c47 Prevent out-of-bounds reads for short FEC packets. by pbos · 10 years ago
  96. 1ca324f Adds UMA histogram for system delay jumps by Bjorn Volcker · 10 years ago
  97. c689124 Simplify OWNERS structure in modules/audio_coding by henrik.lundin · 10 years ago
  98. 9b9f338 Adding Minyue to audio_coding/OWNERS by henrik.lundin · 10 years ago
  99. 9bc2c61 Roll chromium_revision 9729297..fbf756f (335266:336289) by phoglund · 10 years ago
  100. 241338e Added support for keeping a buffer of the previous X seconds, to add to an AcmDump. by Ivo Creusen · 10 years ago