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webrtc / src / db5d016202aab336946b9ca0feadde7c5c171f3f / modules
  1. 6967553 APM-QA Test data generation: environmental noise looped. by Alessio Bazzica · 7 years ago
  2. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  3. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  4. 1d87b0e Create RtcEventLogEncoderLegacy by Elad Alon · 7 years ago
  5. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  6. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  7. d25fa78 Revert "Make rtc_base/refcount.h self contained, not including refcountedobject.h." by Niels Moller · 7 years ago
  8. b7239a9 Make rtc_base/refcount.h self contained, not including refcountedobject.h. by Niels Möller · 7 years ago
  9. 849b3ae Move list of supported H264 codecs from InternalEncoderFactory to h264.h by Magnus Jedvert · 7 years ago
  10. 3c0bbff Remove deprecated mac screencapture code. by Kári Tristan Helgason · 7 years ago
  11. 4db97b9 Enable and update some bit exactness tests by Sam Zackrisson · 7 years ago
  12. 958288a Fix wrap-around logic in ForwardErrorCorrection. by Niels Möller · 7 years ago
  13. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  14. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  15. c007857 AEC3 tunings to increase transparency by Per Åhgren · 7 years ago
  16. 85a11a3 Bounding the AEC3 suppression gain for poorly estimated residual echoes by Per Åhgren · 7 years ago
  17. 707f278 Add RTT to playout delay behind WebRTC-AddRttToPlayoutDelay field trial. by philipel · 7 years ago
  18. 8e56076 LogDelayBasedBweUpdate on detector state change. by philipel · 7 years ago
  19. b378a22 Fix ALR field trial parsing by Erik Språng · 7 years ago
  20. c545daf Make rtp_packet.h public by Elad Alon · 7 years ago
  21. 7dc719a Remove duplicate packet check from webrtc::PacketQueue. by philipel · 7 years ago
  22. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  23. bf35298 Implement temporal layers checkers for vp8 by Ilya Nikolaevskiy · 7 years ago
  24. 884e49f Convert PayloadUnion from a union to a class, step 3 by Karl Wiberg · 7 years ago
  25. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  26. d4a790f Remove AudioCodingModule::IncomingPayload by Henrik Lundin · 7 years ago
  27. a86ac6d Improves UMA stat for built-in AGC monitoring on iOS by henrika · 7 years ago
  28. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  29. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  30. 5bc0229 Injectable APM simulator binary in APM-QA by Alessio Bazzica · 7 years ago
  31. c856dc2 Convert PayloadUnion from a union to a class, step 2 by Karl Wiberg · 7 years ago
  32. 83d3ec1 Convert PayloadUnion from a union to a class, step 1 by Karl Wiberg · 7 years ago
  33. 612f858 Adding test for SingleNalUnit mode by ssilkin · 7 years ago
  34. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  35. a81403f Calculate VP9 references to wrap at kPicIdLength instead of 16 bits. by philipel · 7 years ago
  36. 760c4b4 Trigger rtt and stats update on report block rather than receiver report. by Danil Chapovalov · 7 years ago
  37. 7e9c614 Added configurable offsets to the per-packet overhead in the ANA frame length and bitrate controllers. by ivoc · 7 years ago
  38. a82fcd0 Remove unused mocks of process thread by Danil Chapovalov · 7 years ago
  39. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  40. df5bb65 Prepare to remove ADM APIs that are to be deprecated. by solenberg · 7 years ago
  41. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  42. 9981bd9 Move PacketQueue out of paced_sender.cc to its own packet_queue.{cc,h}. by philipel · 7 years ago
  43. 638200e Add support for SW fallback decoder in VideoProcessor. by Rasmus Brandt · 7 years ago
  44. c9d5b05 Add lock annotations and const declarations to RtpReceiverImpl. by Niels Möller · 7 years ago
  45. ca90a55 audioproc_f with simulated mic analog gain by Alessio Bazzica · 7 years ago
  46. 29accef Export script bug fixed. by Alessio Bazzica · 7 years ago
  47. fe9f222 Reland of Added logging inside AEC3 for render API buffer by Per Åhgren · 7 years ago
  48. bbf389c Delete redundant logic for setting is_first_packet_in_frame by Niels Möller · 7 years ago
  49. 0beac58 Add PostProcessing interface to audio processing module. by Sam Zackrisson · 7 years ago
  50. 5d26edc Total Harmonic Distorsion plus noise (THD+n) score in APM-QA. by alessiob · 7 years ago
  51. a420551 Push back on the video encoder to avoid building queues in the pacer. by philipel · 7 years ago
  52. e19d8bf Modify some rate control and quality thresholds due to flakiness. by asapersson · 7 years ago
  53. 8f1b93c Add more logs in DX capturer by Zijie He · 7 years ago
  54. dccfc40 NetEq: Simplify the dependencies of GetNetworkStatistics by Henrik Lundin · 7 years ago
  55. dec82ab Disable flaky test VideoProcessorIntegrationTestMediaCodec.ForemanCif500kbpsVp8. by Alex Loiko · 7 years ago
  56. 6b3e1a2 Fixes issue in ADM on Mac OSX when audio is renegotiated by henrika · 7 years ago
  57. 599df85 Resolve cyclic dependency in remote bitrate estimator by Danil Chapovalov · 7 years ago
  58. fb08994 Adding time profiling support to AudioFrame by henrika · 7 years ago
  59. 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
  60. ac0a503 NetEq/Stats: Don't let concealed_samples decrease by Henrik Lundin · 7 years ago
  61. b3547fa Revert "Added logging inside AEC3 for render API buffer under/overruns" by Per Åhgren · 8 years ago
  62. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 8 years ago
  63. c3d0da0 Avoids crash in AudioTrack when audio starts in background mode by henrika · 8 years ago
  64. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 8 years ago
  65. 55c7ede VideoProcessorIntegrationTest: Group member variables into two structs containing target/actual rates. by asapersson · 8 years ago
  66. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 8 years ago
  67. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 8 years ago
  68. 92d9dd0 rtp_rtcp_format: Separate public and private source files by Karl Wiberg · 8 years ago
  69. b335e31 This is a rollback of https://chromium-review.googlesource.com/c/external/webrtc/+/616724 by alexnarest · 8 years ago
  70. 080832e Moving Obj-C++ code in desktop_capture_objc. by Mirko Bonadei · 8 years ago
  71. 2572404 Removing useless include_dirs entry. by Mirko Bonadei · 8 years ago
  72. a5f043f Change ForwardErrorCorrection class to accept one received packet at a time. by nisse · 8 years ago
  73. c5267d2 Simplify ReceiveStatistics: merge GetActiveStatisticians into RtcpReportBlocks by Danil Chapovalov · 8 years ago
  74. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 8 years ago
  75. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  76. 4354725 Delete deprecated metod RtpRtcp::SetMaxTransferUnit. by nisse · 8 years ago
  77. 930021d Eliminating the risk of sustained echo during capture data loss in AEC3. by Per Åhgren · 8 years ago
  78. a7567a9 Implement DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper by Zijie He · 8 years ago
  79. 6c17057 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target by Danil Chapovalov · 8 years ago
  80. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 8 years ago
  81. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  82. 262d4ff Added logging inside AEC3 for render API buffer under/overruns by Per Åhgren · 8 years ago
  83. 9a45116 Fix Gn Untracked headers in webrtc/common_audio by charujain · 8 years ago
  84. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  85. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago
  86. 5adc73a by niklase@google.com · 14 years ago
  87. f0a476b Add PictureID and NonReference to codec information by hlundin@google.com · 14 years ago
  88. d0159d8 aec_rdft_128: one entry point for each sign. by cduvivier@google.com · 14 years ago
  89. fae3b31 Optimization/cleanup of 'aec_rfdt' initialization (constants, LUT, ...): by cduvivier@google.com · 14 years ago
  90. 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 · 14 years ago
  91. 98b4ed1 Disabling DEBUG_FILE in the overuse detector by default. by holmer@google.com · 14 years ago
  92. 2b4b7f1 Moving two testfiles, audio coding module. by tlegrand@google.com · 14 years ago
  93. 0adca82 Move iLBC test and reference files to new location. by tlegrand@google.com · 14 years ago
  94. cdc943e VCM: 1. Updating handling of empty packets. 2. Updating JB test. 3. Removing un-used code. by mikhal@google.com · 14 years ago
  95. c137082 Update media_opt_util with frame size parameters. by marpan@google.com · 14 years ago
  96. 6b04739 Route CodecSpecificInfo from encoder to packetizer by hlundin@google.com · 14 years ago
  97. 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 · 14 years ago
  98. 67d7282 Allow the FEC to protect up to maximum #packets (48) if the by marpan@google.com · 14 years ago
  99. 9d94116 Optimization of 'rftbsub': by cduvivier@google.com · 14 years ago
  100. 8ec2231 Add aec_rdft.c to android build by leozwang@google.com · 14 years ago