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webrtc / src / webrtc / 0c1546cd86378cd2b546cd6bc6bef37983e5da8c
  1. 0c1546c audio_coding: remove "main" directory by kjellander · 9 years ago
  2. 6c6de9d Build/use constructormagic.h unconditionally. by Peter Boström · 9 years ago
  3. f9b2421 Standalone denoiser (off by default). by jackychen · 9 years ago
  4. eb6b4f7 Removed api call that will break the upcoming thread checking scheme by peah · 9 years ago
  5. f0ddc91 Roll chromium_revision aa8e58a..664fe1e (361601:361806) by Henrik Kjellander · 9 years ago
  6. 57ea0a4 Strip IP addresses in NDEBUG (release) builds. by Peter Boström · 9 years ago
  7. 5da9b8f Roll chromium_revision 68cf0b8..aa8e58a (361406:361601) by kjellander · 9 years ago
  8. 9c2f41e GetDefaultLocalAddress should return false when the address is invalid by Guo-wei Shieh · 9 years ago
  9. a48ddec Fix fuzzer breakage in Chromium. by Peter Boström · 9 years ago
  10. 5124e1a Move Chromium logging into rtc_base_approved. by Peter Boström · 9 years ago
  11. 4e204c7 Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ ) by kjellander · 9 years ago
  12. ad95edd Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  13. 65c568f Destroy a Connection if a CreatePermission request fails. by deadbeef · 9 years ago
  14. 5bbf7f9 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  15. 7d0b6c1 Removed the aec state as an input parameter to the FilterFar function. by peah · 9 years ago
  16. 825742c A unittest that reports the statistics for the duration of an APM stream processing API call. by peah · 9 years ago
  17. d521902 Move the FEC enabling logic from CodecManager to Rent-A-Codec by kwiberg · 9 years ago
  18. 8af615d Move the stereo-disables-CNG logic from CodecManager to Rent-A-Codec by kwiberg · 9 years ago
  19. c514178 Set mac_deployment_target default to 10.7 by Henrik Kjellander · 9 years ago
  20. 7ad8fc8 Remove frame time scheduing in IncomingVideoStream by qiangchen · 9 years ago
  21. 1e8f5f0 Revert "GetDefaultLocalAddress should return false when the address is invalid" by Guo-wei Shieh · 9 years ago
  22. 64b7954 GetDefaultLocalAddress should return false when the address is invalid by Guo-wei Shieh · 9 years ago
  23. dbb9f9b Move thread_ conditional back under defines. by Peter Boström · 9 years ago
  24. fe2d50c Skip setting thread priorities in NaCl. by Peter Boström · 9 years ago
  25. ebb00d1 Improve documentation for ArrayView by kwiberg · 9 years ago
  26. 60c5d9b Remove duplicated headers after updating downstream code. by kjellander · 9 years ago
  27. 4799095 Work around data race in TransmitMixer. by solenberg · 9 years ago
  28. 827b2d3 Remove VIDEOCODEC_* from engine_configurations.h. by Peter Boström · 9 years ago
  29. 6ec6cfb Inline ConvertToSystemPriority. by Peter Boström · 9 years ago
  30. d419613 Add option to capture to texture in AppRTCDemo for Android. by Per · 9 years ago
  31. a5161c2 First part of the preparatory work before the actual work for solving the ducking problem starts. by peah · 9 years ago
  32. 3db1033 GN: Fix iOS error in audio_device and rtc_base by kjellander · 9 years ago
  33. 9410e01 Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  34. 3adfcea Test case for CL 1437933002. by guoweis · 9 years ago
  35. 6d76de2 Add new method AcmReceiver::last_packet_sample_rate_hz() by henrik.lundin · 9 years ago
  36. 22dae1a Remove the special case for std::vector in rtc::ArrayView by kwiberg · 9 years ago
  37. 0bd578b NetEq: Add new method last_output_sample_rate_hz by henrik.lundin · 9 years ago
  38. ee616be Remove ThreadWrapper::GetThreadId. The method just calls rtc::CurrentThreadId(), which also has a more descriptive name. by Tommi · 9 years ago
  39. fe3129a Implement fuzzing of VP9 depacketization. by Peter Boström · 9 years ago
  40. 2b1da16 Add screenshare perf tests with lossy links by sprang · 9 years ago
  41. 0e2b794 Extract the parameters for the encoder stack from the CodecManager by kwiberg · 9 years ago
  42. fc9226f Request keyframe if too many packets are missing and NACK is disabled. by jbauch · 9 years ago
  43. 8674f94 Remove <iostream> include from file_audio_device.cc by kjellander@webrtc.org · 9 years ago
  44. f1b5d20 RTCP Bye packet moved to own file by danilchap · 9 years ago
  45. 3e38ef9 Increase transport feedback frequency to 20 Hz. by stefan · 9 years ago
  46. 2db00ce Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  47. 8d85ad5 NetEq: Remove overly verbose logging by henrik.lundin · 9 years ago
  48. 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
  49. 775e132 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  50. 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
  51. 0cd2148 Allow pacer to boost bitrate in order to meet time constraints. by sprang · 9 years ago
  52. 732b339 Improved error handling in iOS ADM to avoid race during init by henrika · 9 years ago
  53. 9119471 Avoids hitting DCHECK in OpenSL ES player by henrika · 9 years ago
  54. dba3e45 iOS: Set enable_protobuf=1 by default. by kjellander@webrtc.org · 9 years ago
  55. 568ca73 Add aecdump support to audioproc_f by aluebs · 9 years ago
  56. 0e34004 Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes. by torbjorng · 9 years ago
  57. a67ca87 Remove dead code (we no longer support SILK) by kwiberg · 9 years ago
  58. d0586f8 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot. by ivoc · 9 years ago
  59. dc67f78 Remove build_with_libjingle and exclude failing iOS tests from 'All' target. by kjellander@webrtc.org · 9 years ago
  60. 8448fcf Fix DTLS packet boundary handling in SSLStreamAdapterTests. by jbauch · 9 years ago
  61. 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
  62. 838c3b5 Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  63. 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
  64. 279937b Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  65. 9592d2a modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 9 years ago
  66. 8c330b6 modules/video_coding/utility: Remove include by kjellander@webrtc.org · 9 years ago
  67. 82064b0 modules/video_processing: refactor interface->include + more. by Henrik Kjellander · 9 years ago
  68. 1a4dbb4 WebRTC: Add compability header for video_coding refactoring. by Henrik Kjellander · 9 years ago
  69. fe7633e modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  70. 262cad5 Remove dead code by kwiberg · 9 years ago
  71. 6c7e6c2 Move CNG/RED payload type extraction to Rent-A-Codec by kwiberg · 9 years ago
  72. f7e7c2a Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation by peah · 9 years ago
  73. 2ba093c rtcp::App moved into own file and got Parse function by danilchap · 9 years ago
  74. e12951e So long and thanks for all the code reviews! by andrew · 9 years ago
  75. 82cc96f Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently). by asapersson · 9 years ago
  76. df7f9e5 Fix active tcp port to 9 by Guo-wei Shieh · 9 years ago
  77. 12b7819 Several Tick counter improvements try #2." by thaloun · 9 years ago
  78. 7edbc3a Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc. by davidben · 9 years ago
  79. 2f4c471 Re-apply change https://codereview.webrtc.org/1426673007/ by honghaiz · 9 years ago
  80. b1e1f5d Add OpenSL ES enable setting to AppRTCDemo (part 2). by henrika · 9 years ago
  81. 87c6742 Remove ViEEncoder::ScaleInputImage. by Peter Boström · 9 years ago
  82. 63c862a Unconditionally build VP9 support. by Peter Boström · 9 years ago
  83. 9ab9f46 Add UMA for send bwe and pacer bitrate. by stefan · 9 years ago
  84. 30dd8bd Trace encoding/decoding time in a generic way. by pbos · 9 years ago
  85. 963832b Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant by henrika · 9 years ago
  86. 3c173a9 Adding thread timeout for audio recorer thread in Java by henrika · 9 years ago
  87. 80b7950 Add OpenSL ES enable setting to AppRTCDemo. by glaznev · 9 years ago
  88. 1b5ad57 Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ ) by pbos · 9 years ago
  89. 568fbab Preparational work before introducing the locks in order to harmonize the code: by peah · 9 years ago
  90. 9323716 Applied the render queueing to the agc. by peah · 9 years ago
  91. 2c58add Remove packet initializer in RtpRtcpRtxNackTest. by pbos · 9 years ago
  92. f8290aa Use webrtc/base/logging.h for video coding/processing. by pbos · 9 years ago
  93. f3ae889 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ ) by thaloun · 9 years ago
  94. 1506348 Introduced the render sample queue for the aec and aecm. by peah · 9 years ago
  95. 45d1ec1 Several Tick counter improvements. by Tim Haloun · 9 years ago
  96. b799b43 Fix VP9 support in AppRTCDemo. by Alex Glaznev · 9 years ago
  97. cd66254 common_video: rename interface -> include by kjellander · 9 years ago
  98. 3492dab Create rtc::AtomicInt POD struct. by pbos · 9 years ago
  99. 2778875 Flesh out webrtc/.gitignore by brucedawson · 9 years ago
  100. 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