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webrtc / src / 5085b0ca94c96ebf85fa2631e8f1facdf07d56da / webrtc / modules
  1. 6bf62f7 Avoids java.lang.NullPointerException in WebRtcAudioRecord by henrika · 9 years ago
  2. 4805231 Moved format_macros.h from rtc_base to rtc_base_approved. by ivoc · 9 years ago
  3. 4bc4d27 GN: Fix Windows Clang errors by ehmaldonado · 9 years ago
  4. 3f746ea Fix error when accumulating floats in an int. by maxmorin · 9 years ago
  5. 19319a3 Add missing "//build/config/sanitizers:deps" to executable targets. by ehmaldonado · 9 years ago
  6. 00e45bb Move InsertZeroColumns and CopyColumn to ::internal. by brandtr · 9 years ago
  7. 7a770e0 GN build rules for four audio processing test executables by kwiberg · 9 years ago
  8. 8a6a600 Make neteq_rtpplay parse RTP header extensions by henrik.lundin · 9 years ago
  9. 5f09980 Removed inline definitions and added destructors to fix chromium-style. by aleloi · 9 years ago
  10. 549d80b NetEq: only update current_rtp_payload_type_ when validated by henrik.lundin · 9 years ago
  11. fcada90 Fixing timestamp comparison assert. by deadbeef · 9 years ago
  12. 36a06a9 Increase QP threshold for H.264 encoder QP based scaling. by glaznev · 9 years ago
  13. 5fac3f0 NetEq: Don't check sample rate and frame size upon error by henrik.lundin · 9 years ago
  14. d1a10a0 Make FakeDecodeFromFile handle codec-internal CNG by henrik.lundin · 9 years ago
  15. 28a0ffd GN: Synchronize resources between Android and iOS. by kjellander · 9 years ago
  16. 2ec45b9 Make dependency of audio_device of ApplicationServices explicit. by maxmorin · 9 years ago
  17. 4e7e8d7 Now probe for x3 and x6 of the initial start bitrate and allow for faster receive bitrates when calculating probing estimates. by philipel · 9 years ago
  18. 2c670db Added GN target for webrtc_opus_fec_test. by ivoc · 9 years ago
  19. 98468bb Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ ) by sakal · 9 years ago
  20. 538b560 GN build rules for four audio processing test executables by kwiberg · 9 years ago
  21. 0561bdf Only use payload size within the know send/receive interval for probing calculations. by philipel · 9 years ago
  22. 619a211 iLBC: Handle a case of bad input data by kwiberg · 9 years ago
  23. 0aa9d18 Set send side bitrate estimate on successful probing attempt. by philipel · 9 years ago
  24. e51b41a Added GN target for isac_test. by ivoc · 9 years ago
  25. 5d167d6 Removals and renamings in the new audio mixer. by aleloi · 9 years ago
  26. 30be5d7 Updated mixer unittests and fixed a related bug in the new mixer. by aleloi · 9 years ago
  27. 616df1e Added a level indicator to new mixer. by aleloi · 9 years ago
  28. f99a9de ProbingEstimator: Erase history based on time threshold by Irfan Sheriff · 9 years ago
  29. a246cfb Don't include RTP headers in send-side BWE. by Stefan Holmer · 9 years ago
  30. 9a11784 Migrated GN target :g722_test by aleloi · 9 years ago
  31. 16f55a1 Migrated GN target :g711_test by aleloi · 9 years ago
  32. 2e48646 RTC_CHECK and RTC_DCHECK macros for C by kwiberg · 9 years ago
  33. d8dd190 GN: Fix test_support_unittests and MIPS compile issue. by kjellander · 9 years ago
  34. b3f1c5d Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 9 years ago
  35. 6c46eaa Add gtest as a dependency for neteq_quality_test_support. by ehmaldonado · 9 years ago
  36. d48717b Fix issue where the number of packets reported in tests/simulations sometimes are negative. by stefan · 9 years ago
  37. 4ec01d9 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 9 years ago
  38. 853ecb2 Style cleanup in UpdateTmmbr: by danilchap · 9 years ago
  39. 7f82fc9 WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows) by kwiberg · 9 years ago
  40. 642e3bc [rtcp] TransportFeedback adjusted to match other rtcp packets: by danilchap · 9 years ago
  41. 4981051 [Reland] Cleanup of the AudioDeviceBuffer class. by henrika · 9 years ago
  42. 83d79cd Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ ) by kjellander · 9 years ago
  43. e5b4141 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData by danilchap · 9 years ago
  44. abcc3de Add pps id and sps id parsing to the h.264 depacketizer. by stefan · 9 years ago
  45. 8177452 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers by magjed · 9 years ago
  46. d7a89db Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ ) by henrika · 9 years ago
  47. cf327b4 Cleanup of the AudioDeviceBuffer class. by henrika · 9 years ago
  48. da161d7 Reformat rtcp_receiver git cl format --full by danilchap · 9 years ago
  49. 861da3c Refactor neteq_test_support. by ehmaldonado · 9 years ago
  50. bcba64a GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. by ehmaldonado · 9 years ago
  51. b5b3090 Corrected the testvectors for the level controller by peah · 9 years ago
  52. 8df4d0e Add playout_delay_oracle_unittest as gn target by isheriff · 9 years ago
  53. 3a11933 Remove audio_device_test_func. by maxmorin · 9 years ago
  54. 644fa96 Added recording of the configuration for the AudioFrame API call by peah · 9 years ago
  55. 2b61639 Remove TMMBRSet class by danilchap · 9 years ago
  56. 38d840c NetEq: Changing checked_cast to saturated_cast by henrik.lundin · 9 years ago
  57. e9a6acf Added missing unittest to the modules/BUILD.gn build file by peah · 9 years ago
  58. cb2d701 Add kjellander as BUILD.gn OWNER in webrtc/modules by kjellander · 9 years ago
  59. 71fead2 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ ) by danilchap · 9 years ago
  60. d4e9f62 Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats. by ossu · 9 years ago
  61. 235020d Roll chromium_revision 915e47250f..e3860bd297 (412201:412289) by magjed · 9 years ago
  62. 6a35590 Add code for dummy file audio to fallback to dummy audio. by noahric · 9 years ago
  63. d8a72f0 Close input file in FileAudioDevice::StopRecording. by noahric · 9 years ago
  64. d22854b FilePlayer: Remove unused default values for arguments by kwiberg · 9 years ago
  65. 4a42900 Removes redundant log warning in WebRtcAudioManager. by henrika · 9 years ago
  66. 86c9694 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ ) by danilchap · 9 years ago
  67. 5a25d95 FileRecorder + FilePlayer: Let Create functions return unique_ptr by kwiberg · 9 years ago
  68. 4466782 StartTimestamp generated randomly in RtpSender constructor by Danil Chapovalov · 9 years ago
  69. 144dd27 FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files by kwiberg · 9 years ago
  70. c54071d WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. by ossu · 9 years ago
  71. eb680ea CongestionController::SetBweBitrates may now trigger probing. by philipel · 9 years ago
  72. c594aa61 Add a gyp/gn option to use dummy audio file devices. by noahric · 9 years ago
  73. 49c01d7 Currently there is not way to programmically test whether a ScreenCapturer by zijiehe · 9 years ago
  74. 287e548 Cleanup RtcpReceiver::TMMBRReceived function by danilchap · 9 years ago
  75. a06ce49 Run "git cl format" on some files before I start to modify them by kwiberg · 9 years ago
  76. 90920d5 GN: Enable msse2 flag in Mac. by ehmaldonado · 9 years ago
  77. 9d7eb13 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ ) by kwiberg · 9 years ago
  78. 427ce3d Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 9 years ago
  79. 2f69ce9 Cleaned out candidateSet member from TMMBRHelp class by danilchap · 9 years ago
  80. 6910537 Add gn target for audio_device_tests. by maxmorin · 9 years ago
  81. 70f866c Added new mixer to |check_targets| in .gn and fixed include/depend errors. by aleloi · 9 years ago
  82. 7522a28 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe. by philipel · 9 years ago
  83. b7186d0 Migrated GN target :isac_fix_test by aleloi · 9 years ago
  84. 88e31a3 Fix warnings, simplify ADM. by maxmorin · 9 years ago
  85. 82dda1a [WebRTC] Disable DirectX capturer tests if the system does not support it. by zijiehe · 9 years ago
  86. e1b4d24 Skip AUD while extracting SPS and PPS on iOS. by jianjun.zhu · 9 years ago
  87. 6c687e7 Make prior H264 QP adjustments iOS specific. by tkchin · 9 years ago
  88. 3473288 Remove VERBOSE logs in (android) audio device code. by noahric · 9 years ago
  89. b1e6611 GN: Fix audio_decoder_unittests for android. by ehmaldonado · 9 years ago
  90. 4a1ec1e Added ProbeBitrate(bitrate_bps, num_probes) to BitrateProber. by philipel · 9 years ago
  91. 1aee0b5 Remove old methods in AudioTransport, make it pass a gn build by maxmorin · 9 years ago
  92. c8c71f4 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ ) by kwiberg · 9 years ago
  93. dc65ea2 Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 9 years ago
  94. f96c51a GN: Add video_capture_tests for Mac by kjellander · 9 years ago
  95. 2ab1da7 Revert of Added new mixer to |check_targets| in .gn and fixed include/depend errors. (patchset #1 id:1 of https://codereview.webrtc.org/2234293002/ ) by olka · 9 years ago
  96. d700bef Added new mixer to |check_targets| in .gn and fixed include/depend errors. by aleloi · 9 years ago
  97. 963be23 RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods by kwiberg · 9 years ago
  98. 5bcc00e Changed folder structure in new mixer and fixed simple lint errors. by aleloi · 9 years ago
  99. 714dd4e GN: Update tests to have the correct shard timeout value on Android. by sakal · 9 years ago
  100. 64a7eab Update tests and DTX check for Opus 1.1.3. by flim · 9 years ago