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webrtc / src / 4c17abe35e0ffaea71dd3c2f5f67c3ddf49a6ef3 / webrtc / modules
  1. 4c17abe Add DesktopCapturer::Result::MAX_VALUE by Sergey Ulanov · 9 years ago
  2. a6219cc FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  3. ceb9d0c Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz by kwiberg · 9 years ago
  4. 6808419 iSAC decoder: Remove obsolete TODO by kwiberg · 9 years ago
  5. 71ee44c This cl: by perkj · 9 years ago
  6. 17c3cdd Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ ) by tommi · 9 years ago
  7. 2cc8baa Adjust the amount of VP8 encoder threads for Android builds. by Alex Glaznev · 9 years ago
  8. 5aaa9fa Remove thread_checker in playout_delay_oracle by isheriff · 9 years ago
  9. b1963b4 Reland of Re-enable UBsan on AGC. by minyue · 9 years ago
  10. 1c7eef6 Split IncomingVideoStream into two implementations, with smoothing and without. by tommi · 9 years ago
  11. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  12. 880ffeb Optimize the repeated calls to AudioEffect.queryEffects() on Android by skvlad · 9 years ago
  13. abfdb53 Fixed partially out of screen window capture in unix by gyzhou · 9 years ago
  14. 718a763 Refactor scaling. by Niels Möller · 9 years ago
  15. be99ab9 Remove unnecessary redefinition of PacketLists in rtp_fec_unittest. by Rasmus Brandt · 9 years ago
  16. fb11424 GN: Add modules_unittests by kjellander · 9 years ago
  17. 82a9449 GN: Add rtc_media_unittests by kjellander · 9 years ago
  18. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  19. 51e6030 Update RateStatistics to handle too-little-data case. by Erik Språng · 9 years ago
  20. 602844a Delete some unused header files. by nisse · 9 years ago
  21. 94cee31 GN: Enable api,media,pc and p2p for the 'webrtc' target. by kjellander · 9 years ago
  22. 2b9423f Revert of Re-enable UBsan on AGC. (patchset #8 id:300001 of https://codereview.webrtc.org/2003623003/ ) by Åsa Persson · 9 years ago
  23. d4070c6 GN: Fix Chromium breakage for remote_bitrate_estimator by Henrik Kjellander · 9 years ago
  24. 5c1d043 Fix GYP/GN for webrtc/modules/remote_bitrate_estimator by kjellander · 9 years ago
  25. f2a1c89 Add r-value constructor for RefCountedObject. by sergeyu · 9 years ago
  26. bde418d Renamed video_coding/packet_buffer_unittest.cc. by philipel · 9 years ago
  27. 2019afd Replaced ACCESS_ON alias with GUARDED_BY macros by danilchap · 9 years ago
  28. e8f8f60 Only update Intelligibility Enhancer gains every 10 chunks by aluebs · 9 years ago
  29. 9195186 NetEq: Rename Nack to NackTracker to avoid name collisions in GN by henrik.lundin · 9 years ago
  30. bbe4233 Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files by peah · 9 years ago
  31. a107402 Fix UBSan errors (signed integer overflow) by kwiberg · 9 years ago
  32. 3bcedd3 GN: Add SDK tests to rtc_unittests. by kjellander · 9 years ago
  33. 6b4b5f3 Add sender controlled playout delay limits by isheriff · 9 years ago
  34. 5d91028 Use std::unique_ptr<> to pass frame ownership in DesktopCapturer impls. by sergeyu · 9 years ago
  35. 6ebdf6b Fix issue with parsing of incorrect (empty) Stap-A H264 NAL units. by Erik Språng · 9 years ago
  36. c88f558 Fix Android audio playback mute. by Alex Glaznev · 9 years ago
  37. 080be51 Make WebRTCAudioTrack class public. by Alex Glaznev · 9 years ago
  38. 7bf939c Check for out-of-bounds access on |kIntrpCoef|. by mbarbella · 9 years ago
  39. 647998c Use picture id to check continuity between frames. by philipel · 9 years ago
  40. 293c86d Re-enable UBsan on AGC. by minyue · 9 years ago
  41. b4c7b83 Revert of Use std::unique_ptr<> to pass frame ownership in DesktopCapturer impls. (patchset #7 id:140001 of https://codereview.webrtc.org/1988783003/ ) by sergeyu · 9 years ago
  42. 4a627a8 Use std::unique_ptr<> to pass frame ownership in DesktopCapturer impls. by sergeyu · 9 years ago
  43. b50e845 Adds WebRtcAudioTrack.setSpeakerMute() API by henrika · 9 years ago
  44. fc715f5 DCHECK that the red payload type doesn't have invalid values when FEC is enabled. by stefan · 9 years ago
  45. 1a93cde BitrateProber now correctly change state to kWait when the last probing by philipel · 9 years ago
  46. 846b2d9 Reduce logging frequency in bwe simulations. by stefan · 9 years ago
  47. 8f4c77f Always send RED headers if configured. by stefan · 9 years ago
  48. 72d41aa Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ ) by guidou · 9 years ago
  49. 2221e1c Update the BWE when the network route changes. by honghaiz · 9 years ago
  50. 69b332d Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender. by Per · 9 years ago
  51. 58d4fe7 Reduce VT keyframe frequency. by Peter Bostrom · 9 years ago
  52. b684bd1 [rtcp] ExtendedJitterReports::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  53. f882880 AudioDecoder: Document that the sample rate and number of channels is constant by kwiberg · 9 years ago
  54. abe95ba AudioDecoderIsacT: Require caller to always specify sample rate by kwiberg · 9 years ago
  55. 52033d6 Add H264 bitstream rewriting to limit frame reordering marker in header by sprang · 9 years ago
  56. bac0412 GN: Add system_wrappers_unittests, tools and tools_unittests by kjellander · 9 years ago
  57. 8f4419b GN: Replace Windows suppressions of warning 4267 with config. by kjellander · 9 years ago
  58. 69adc9c Delete unused code in webrtc_libyuv.cc. by nisse · 9 years ago
  59. a1ed0b3 Revert "Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )" by philipel · 9 years ago
  60. 46948c1 Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ ) by philipel · 9 years ago
  61. 5be28c8 Propagate probing cluster id to SendTimeHistory, both for packets and padding. by philipel · 9 years ago
  62. 521f7a8 Moves ownership of OpenSL engine object to Android audio manager with the goal of adding support for OpenSL ES based audio capture. by henrika · 9 years ago
  63. c0f2dcf NetEq decoder database: Don't keep track of sample rate for builtin decoders by kwiberg · 9 years ago
  64. 938c5dd [rtcp] RapidResyncRequest::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  65. 208d198 Rename APK tests workaround to make it more generic. by kjellander · 9 years ago
  66. 6c2eab3 AudioDecoder: New method SampleRateHz, + implementations for our codecs by kwiberg · 9 years ago
  67. 946f36e Added diagnostic AEC debug logpoints for the purpose by peah · 9 years ago
  68. 0fc37d7 [rtcp] Fir/Sli/Rpsi updated not to use RTCPUtility by danilchap · 9 years ago
  69. 2d3aae5 [rtcp] ExtendedReports::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  70. 073eb7e Updated #if in IsH264CodecSupported to match H264[En/De]coder::Create. by hbos · 9 years ago
  71. 435f98b Reland of Change ProcessThread's task type to be the one from TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/2020783003/ ) by tommi · 9 years ago
  72. 6414551 Revert of Change ProcessThread's task type to be the one from TaskQueue. (patchset #3 id:80001 of https://codereview.webrtc.org/2016043003/ ) by tommi · 9 years ago
  73. 400a276 Change ProcessThread's task type to be the one from TaskQueue. by Tommi · 9 years ago
  74. 90edc65 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ ) by Tommi · 9 years ago
  75. 5771beb Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ ) by Tommi · 9 years ago
  76. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  77. b4ff7a7 Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2014183003/ ) by tommi · 9 years ago
  78. b7318f1 Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006313009/ ) by tommi · 9 years ago
  79. 303d3e1 Fixing neteq_rtpplay by henrik.lundin · 9 years ago
  80. 73257d1 Reduce flakiness in IvfFileWriter tests. by sprang · 9 years ago
  81. 6ce1adc [rtcp] Tmmbn/Tmmbr Parse updated not to use RTCPUtility by danilchap · 9 years ago
  82. 4f6c2b6 Fix UBSan errors (left shift of negative value) by kwiberg · 9 years ago
  83. 6895d8c Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2009253004/ ) by kjellander · 9 years ago
  84. ba189cc Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2006243002/ ) by tommi · 9 years ago
  85. 080a1e3 Fix iOS GN build and cleanup system_wrappers by kjellander · 9 years ago
  86. a4463bc Report framedrops outside libvpx to QualityScaler. by Peter Boström · 9 years ago
  87. 612c25e NetEq: Fix stats counting in muted mode by henrik.lundin · 9 years ago
  88. e352578 Moved injection of AudioDecoderFactory into voe::Channel. by ossu · 9 years ago
  89. 5777910 Forward Encode failure codes from sub encoders. by noahric · 9 years ago
  90. 1f0ad10 Adds support for detection of pro-audio support on Android. by henrika · 9 years ago
  91. c00687f Add an option to disable built-in AEC to AppRTC Android Demo by sakal · 9 years ago
  92. 46ba49c Let PacketSource::NextPacket() return an std::unique_ptr by henrik.lundin · 9 years ago
  93. 60a189f This CL updates and extends the audioproc_f command line by peah · 9 years ago
  94. 13deaad TMMBRHelp moved from member object/base class to stack object, by danilchap · 9 years ago
  95. fb98b9e Revert of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #6 id:100001 of https://codereview.webrtc.org/2007563002/ ) by tommi · 9 years ago
  96. ce5570e Move neteq_rtpplay.cc inside webrtc::test namespace by henrik.lundin · 9 years ago
  97. 58530ed Updating APM unittests on the echo metrics. by minyue · 9 years ago
  98. d36df89 Adding a some checks and switching out a few assert for RTC_[D]CHECK. by tommi · 9 years ago
  99. e23e760 Integrate QualityScaler into VideoToolboxEncoder. by Peter Boström · 9 years ago
  100. cc1543a Move H264BitstreamParser to video_coding. by Peter Boström · 9 years ago