1. dbb9f9b Move thread_ conditional back under defines. by Peter Boström · 9 years ago
  2. fe2d50c Skip setting thread priorities in NaCl. by Peter Boström · 9 years ago
  3. ebb00d1 Improve documentation for ArrayView by kwiberg · 9 years ago
  4. 60c5d9b Remove duplicated headers after updating downstream code. by kjellander · 9 years ago
  5. 4799095 Work around data race in TransmitMixer. by solenberg · 9 years ago
  6. 827b2d3 Remove VIDEOCODEC_* from engine_configurations.h. by Peter Boström · 9 years ago
  7. 6ec6cfb Inline ConvertToSystemPriority. by Peter Boström · 9 years ago
  8. d419613 Add option to capture to texture in AppRTCDemo for Android. by Per · 9 years ago
  9. a5161c2 First part of the preparatory work before the actual work for solving the ducking problem starts. by peah · 9 years ago
  10. 3db1033 GN: Fix iOS error in audio_device and rtc_base by kjellander · 9 years ago
  11. 9410e01 Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  12. 3adfcea Test case for CL 1437933002. by guoweis · 9 years ago
  13. 6d76de2 Add new method AcmReceiver::last_packet_sample_rate_hz() by henrik.lundin · 9 years ago
  14. 22dae1a Remove the special case for std::vector in rtc::ArrayView by kwiberg · 9 years ago
  15. 0bd578b NetEq: Add new method last_output_sample_rate_hz by henrik.lundin · 9 years ago
  16. ee616be Remove ThreadWrapper::GetThreadId. The method just calls rtc::CurrentThreadId(), which also has a more descriptive name. by Tommi · 9 years ago
  17. fe3129a Implement fuzzing of VP9 depacketization. by Peter Boström · 9 years ago
  18. 2b1da16 Add screenshare perf tests with lossy links by sprang · 9 years ago
  19. 0e2b794 Extract the parameters for the encoder stack from the CodecManager by kwiberg · 9 years ago
  20. fc9226f Request keyframe if too many packets are missing and NACK is disabled. by jbauch · 9 years ago
  21. 8674f94 Remove <iostream> include from file_audio_device.cc by kjellander@webrtc.org · 9 years ago
  22. f1b5d20 RTCP Bye packet moved to own file by danilchap · 9 years ago
  23. 3e38ef9 Increase transport feedback frequency to 20 Hz. by stefan · 9 years ago
  24. 2db00ce Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  25. 8d85ad5 NetEq: Remove overly verbose logging by henrik.lundin · 9 years ago
  26. 697d03b Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  27. 775e132 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  28. ecbf8f5 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  29. 0cd2148 Allow pacer to boost bitrate in order to meet time constraints. by sprang · 9 years ago
  30. 732b339 Improved error handling in iOS ADM to avoid race during init by henrika · 9 years ago
  31. 9119471 Avoids hitting DCHECK in OpenSL ES player by henrika · 9 years ago
  32. dba3e45 iOS: Set enable_protobuf=1 by default. by kjellander@webrtc.org · 9 years ago
  33. 568ca73 Add aecdump support to audioproc_f by aluebs · 9 years ago
  34. 0e34004 Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes. by torbjorng · 9 years ago
  35. a67ca87 Remove dead code (we no longer support SILK) by kwiberg · 9 years ago
  36. d0586f8 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot. by ivoc · 9 years ago
  37. dc67f78 Remove build_with_libjingle and exclude failing iOS tests from 'All' target. by kjellander@webrtc.org · 9 years ago
  38. 8448fcf Fix DTLS packet boundary handling in SSLStreamAdapterTests. by jbauch · 9 years ago
  39. 0f714d5 Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ ) by henrika · 9 years ago
  40. 838c3b5 Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  41. 0eaad21 Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 9 years ago
  42. 279937b Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  43. 9592d2a modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 9 years ago
  44. 8c330b6 modules/video_coding/utility: Remove include by kjellander@webrtc.org · 9 years ago
  45. 82064b0 modules/video_processing: refactor interface->include + more. by Henrik Kjellander · 9 years ago
  46. 1a4dbb4 WebRTC: Add compability header for video_coding refactoring. by Henrik Kjellander · 9 years ago
  47. fe7633e modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  48. 262cad5 Remove dead code by kwiberg · 9 years ago
  49. 6c7e6c2 Move CNG/RED payload type extraction to Rent-A-Codec by kwiberg · 9 years ago
  50. f7e7c2a Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation by peah · 9 years ago
  51. 2ba093c rtcp::App moved into own file and got Parse function by danilchap · 9 years ago
  52. e12951e So long and thanks for all the code reviews! by andrew · 9 years ago
  53. 82cc96f Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently). by asapersson · 9 years ago
  54. df7f9e5 Fix active tcp port to 9 by Guo-wei Shieh · 9 years ago
  55. 12b7819 Several Tick counter improvements try #2." by thaloun · 9 years ago
  56. 7edbc3a Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc. by davidben · 9 years ago
  57. 2f4c471 Re-apply change https://codereview.webrtc.org/1426673007/ by honghaiz · 9 years ago
  58. b1e1f5d Add OpenSL ES enable setting to AppRTCDemo (part 2). by henrika · 9 years ago
  59. 87c6742 Remove ViEEncoder::ScaleInputImage. by Peter Boström · 9 years ago
  60. 63c862a Unconditionally build VP9 support. by Peter Boström · 9 years ago
  61. 9ab9f46 Add UMA for send bwe and pacer bitrate. by stefan · 9 years ago
  62. 30dd8bd Trace encoding/decoding time in a generic way. by pbos · 9 years ago
  63. 963832b Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant by henrika · 9 years ago
  64. 3c173a9 Adding thread timeout for audio recorer thread in Java by henrika · 9 years ago
  65. 80b7950 Add OpenSL ES enable setting to AppRTCDemo. by glaznev · 9 years ago
  66. 1b5ad57 Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ ) by pbos · 9 years ago
  67. 568fbab Preparational work before introducing the locks in order to harmonize the code: by peah · 9 years ago
  68. 9323716 Applied the render queueing to the agc. by peah · 9 years ago
  69. 2c58add Remove packet initializer in RtpRtcpRtxNackTest. by pbos · 9 years ago
  70. f8290aa Use webrtc/base/logging.h for video coding/processing. by pbos · 9 years ago
  71. f3ae889 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) by thaloun · 9 years ago
  72. 1506348 Introduced the render sample queue for the aec and aecm. by peah · 9 years ago
  73. 45d1ec1 Several Tick counter improvements. by Tim Haloun · 9 years ago
  74. b799b43 Fix VP9 support in AppRTCDemo. by Alex Glaznev · 9 years ago
  75. cd66254 common_video: rename interface -> include by kjellander · 9 years ago
  76. 3492dab Create rtc::AtomicInt POD struct. by pbos · 9 years ago
  77. 2778875 Flesh out webrtc/.gitignore by brucedawson · 9 years ago
  78. 03d4810 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  79. 9b5cde7 Adding stddef.h to opus_inst.h. by minyue · 9 years ago
  80. ffe1ce0 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  81. 3c69ade Move CNG and RED management into the Rent-A-Codec by kwiberg · 9 years ago
  82. 74bf484 Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ ) by tommi · 9 years ago
  83. 273352e Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 9 years ago
  84. 62f8929 Fix a data race in the thread unit tests. by nisse · 9 years ago
  85. 2533f78 Add limit for minimum number of required samples before recording input and sent framerate stats. by asapersson · 9 years ago
  86. 3ecbb85 Do not delete the turn port entry right away when the respective connection is deleted. by honghaiz · 9 years ago
  87. 124daee cleanup: get rid of basicdefs.h include by tfarina · 9 years ago
  88. 41f42f6 Fix flaky tests by honghaiz · 9 years ago
  89. ce58e44 rtcp::Ij renamed to rtcp::ExtendedJitterReport by danilchap · 9 years ago
  90. 516dcbf Remove scoped_ptrs for VCM sender_ and receiver_. by pbos · 9 years ago
  91. 2950018 rtcp::ReportBlock refactored to contain parsing by danilchap · 9 years ago
  92. 78b51e8 Remove BitrateController dependency fromVideoReceiveStream. by mflodman · 9 years ago
  93. 5303026 Rename screenshare test. by philipel · 9 years ago
  94. 1b78cc3 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 9 years ago
  95. aeae9d6 Do not delete a connection until it has not received anything for 30 seconds. by Honghai Zhang · 9 years ago
  96. 3347b88 Schedule a CreatePermissionRequest after the success of a previous request by Honghai Zhang · 9 years ago
  97. a9c3720 Introduces Android API level linting, fixes all current API lint errors. by Patrik Höglund · 9 years ago
  98. 73807f4 Re-add a thread check in Call::Call that was removed by mistake in a rebase. by solenberg · 9 years ago
  99. e22c7db Remove webrtc/modules/video_{capture,render}/include by Henrik Kjellander · 9 years ago
  100. cebb833 OpenSL ES stability improvements. by henrika · 9 years ago