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gerrit-public.fairphone.software / platform / external / webrtc / 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30 / modules
  1. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  2. c007857 AEC3 tunings to increase transparency by Per Åhgren · 7 years ago
  3. 85a11a3 Bounding the AEC3 suppression gain for poorly estimated residual echoes by Per Åhgren · 7 years ago
  4. 707f278 Add RTT to playout delay behind WebRTC-AddRttToPlayoutDelay field trial. by philipel · 7 years ago
  5. 8e56076 LogDelayBasedBweUpdate on detector state change. by philipel · 7 years ago
  6. b378a22 Fix ALR field trial parsing by Erik Språng · 7 years ago
  7. c545daf Make rtp_packet.h public by Elad Alon · 7 years ago
  8. 7dc719a Remove duplicate packet check from webrtc::PacketQueue. by philipel · 7 years ago
  9. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  10. bf35298 Implement temporal layers checkers for vp8 by Ilya Nikolaevskiy · 7 years ago
  11. 884e49f Convert PayloadUnion from a union to a class, step 3 by Karl Wiberg · 7 years ago
  12. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  13. d4a790f Remove AudioCodingModule::IncomingPayload by Henrik Lundin · 7 years ago
  14. a86ac6d Improves UMA stat for built-in AGC monitoring on iOS by henrika · 7 years ago
  15. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  16. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  17. 5bc0229 Injectable APM simulator binary in APM-QA by Alessio Bazzica · 7 years ago
  18. c856dc2 Convert PayloadUnion from a union to a class, step 2 by Karl Wiberg · 7 years ago
  19. 83d3ec1 Convert PayloadUnion from a union to a class, step 1 by Karl Wiberg · 7 years ago
  20. 612f858 Adding test for SingleNalUnit mode by ssilkin · 7 years ago
  21. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  22. a81403f Calculate VP9 references to wrap at kPicIdLength instead of 16 bits. by philipel · 7 years ago
  23. 760c4b4 Trigger rtt and stats update on report block rather than receiver report. by Danil Chapovalov · 7 years ago
  24. 7e9c614 Added configurable offsets to the per-packet overhead in the ANA frame length and bitrate controllers. by ivoc · 7 years ago
  25. a82fcd0 Remove unused mocks of process thread by Danil Chapovalov · 7 years ago
  26. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  27. df5bb65 Prepare to remove ADM APIs that are to be deprecated. by solenberg · 7 years ago
  28. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  29. 9981bd9 Move PacketQueue out of paced_sender.cc to its own packet_queue.{cc,h}. by philipel · 7 years ago
  30. 638200e Add support for SW fallback decoder in VideoProcessor. by Rasmus Brandt · 7 years ago
  31. c9d5b05 Add lock annotations and const declarations to RtpReceiverImpl. by Niels Möller · 7 years ago
  32. ca90a55 audioproc_f with simulated mic analog gain by Alessio Bazzica · 7 years ago
  33. 29accef Export script bug fixed. by Alessio Bazzica · 7 years ago
  34. fe9f222 Reland of Added logging inside AEC3 for render API buffer by Per Åhgren · 7 years ago
  35. bbf389c Delete redundant logic for setting is_first_packet_in_frame by Niels Möller · 7 years ago
  36. 0beac58 Add PostProcessing interface to audio processing module. by Sam Zackrisson · 7 years ago
  37. 5d26edc Total Harmonic Distorsion plus noise (THD+n) score in APM-QA. by alessiob · 7 years ago
  38. a420551 Push back on the video encoder to avoid building queues in the pacer. by philipel · 7 years ago
  39. e19d8bf Modify some rate control and quality thresholds due to flakiness. by asapersson · 7 years ago
  40. 8f1b93c Add more logs in DX capturer by Zijie He · 7 years ago
  41. dccfc40 NetEq: Simplify the dependencies of GetNetworkStatistics by Henrik Lundin · 7 years ago
  42. dec82ab Disable flaky test VideoProcessorIntegrationTestMediaCodec.ForemanCif500kbpsVp8. by Alex Loiko · 7 years ago
  43. 6b3e1a2 Fixes issue in ADM on Mac OSX when audio is renegotiated by henrika · 7 years ago
  44. 599df85 Resolve cyclic dependency in remote bitrate estimator by Danil Chapovalov · 7 years ago
  45. fb08994 Adding time profiling support to AudioFrame by henrika · 7 years ago
  46. e21be1d Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) by philipel · 7 years ago
  47. ac0a503 NetEq/Stats: Don't let concealed_samples decrease by Henrik Lundin · 7 years ago
  48. b3547fa Revert "Added logging inside AEC3 for render API buffer under/overruns" by Per Åhgren · 7 years ago
  49. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  50. c3d0da0 Avoids crash in AudioTrack when audio starts in background mode by henrika · 7 years ago
  51. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  52. 55c7ede VideoProcessorIntegrationTest: Group member variables into two structs containing target/actual rates. by asapersson · 7 years ago
  53. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  54. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  55. 92d9dd0 rtp_rtcp_format: Separate public and private source files by Karl Wiberg · 7 years ago
  56. b335e31 This is a rollback of https://chromium-review.googlesource.com/c/external/webrtc/+/616724 by alexnarest · 7 years ago
  57. 080832e Moving Obj-C++ code in desktop_capture_objc. by Mirko Bonadei · 7 years ago
  58. 2572404 Removing useless include_dirs entry. by Mirko Bonadei · 7 years ago
  59. a5f043f Change ForwardErrorCorrection class to accept one received packet at a time. by nisse · 7 years ago
  60. c5267d2 Simplify ReceiveStatistics: merge GetActiveStatisticians into RtcpReportBlocks by Danil Chapovalov · 7 years ago
  61. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  62. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  63. 4354725 Delete deprecated metod RtpRtcp::SetMaxTransferUnit. by nisse · 7 years ago
  64. 930021d Eliminating the risk of sustained echo during capture data loss in AEC3. by Per Åhgren · 7 years ago
  65. a7567a9 Implement DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper by Zijie He · 7 years ago
  66. 6c17057 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target by Danil Chapovalov · 7 years ago
  67. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 7 years ago
  68. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  69. 262d4ff Added logging inside AEC3 for render API buffer under/overruns by Per Åhgren · 7 years ago
  70. 9a45116 Fix Gn Untracked headers in webrtc/common_audio by charujain · 7 years ago
  71. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  72. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago
  73. 5adc73a by niklase@google.com · 13 years ago
  74. f0a476b Add PictureID and NonReference to codec information by hlundin@google.com · 13 years ago
  75. d0159d8 aec_rdft_128: one entry point for each sign. by cduvivier@google.com · 13 years ago
  76. fae3b31 Optimization/cleanup of 'aec_rfdt' initialization (constants, LUT, ...): by cduvivier@google.com · 13 years ago
  77. 7c4469b Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. by ajm@google.com · 13 years ago
  78. 98b4ed1 Disabling DEBUG_FILE in the overuse detector by default. by holmer@google.com · 13 years ago
  79. 2b4b7f1 Moving two testfiles, audio coding module. by tlegrand@google.com · 13 years ago
  80. 0adca82 Move iLBC test and reference files to new location. by tlegrand@google.com · 13 years ago
  81. cdc943e VCM: 1. Updating handling of empty packets. 2. Updating JB test. 3. Removing un-used code. by mikhal@google.com · 13 years ago
  82. c137082 Update media_opt_util with frame size parameters. by marpan@google.com · 13 years ago
  83. 6b04739 Route CodecSpecificInfo from encoder to packetizer by hlundin@google.com · 13 years ago
  84. b5427cb Changing JPEG API to to accept rawImage and encodedImage; moved video_image.h from modules/video_coding/codecs to common_video/interface, and some general re-write to JPEG, especially with regard to memory handling. Required VCM/ViE changes are also included. by mikhal@google.com · 13 years ago
  85. 67d7282 Allow the FEC to protect up to maximum #packets (48) if the by marpan@google.com · 13 years ago
  86. 9d94116 Optimization of 'rftbsub': by cduvivier@google.com · 13 years ago
  87. 8ec2231 Add aec_rdft.c to android build by leozwang@google.com · 13 years ago
  88. 20cb6b6 Optimization of 'rftfsub': by cduvivier@google.com · 13 years ago
  89. 190d087 Remove included header files on that unit_test is not dependent, correct error in last CL by leozwang@google.com · 13 years ago
  90. 6fb5d19 Add Android.mk for apm unit test and make it compile on android by leozwang@google.com · 13 years ago
  91. 21a4405 VPLIB/Interpolation - Delete decode buffer only if too small, this required an API change. In addition, done some clean up and updated test and related code in VCM. by mikhal@google.com · 13 years ago
  92. 1eccf7d Some code cleanup for rtp_sender_video.cc. by marpan@google.com · 13 years ago
  93. e02b57e Updates to qm_select: Function to update content state, and function for FEC rate adjustment. by marpan@google.com · 13 years ago
  94. 6cc3f00 Include forward_error_correction_internal.cc which was added in #93 to android build by leozwang@google.com · 13 years ago
  95. 181f543 AEC specific version of " Real Discrete Fourier Transform". by cduvivier@google.com · 13 years ago
  96. 3ad9c18 Update on content metrics: by marpan@google.com · 13 years ago
  97. 0d7e5bc Fix bug on key frame boost allocation, and some update/cleanup to same function. by marpan@google.com · 13 years ago
  98. 3c45dfd Fixes valgrind warnings in the rtp_rtcp module. by hellner@google.com · 13 years ago
  99. 95fa29e Creating a new directory for test data files, and moving audio_processing files there. by ajm@google.com · 13 years ago
  100. 4bf9c0b Adds sanity checks related to IAudioCaptureClient::GetBuffer. by henrika@google.com · 13 years ago