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gerrit-public.fairphone.software / platform / external / webrtc / 11d8176cb3383a2f96e118ff054e92e97a8d9db4
  1. 11d8176 Move updating nack bitrate inside UpdateNACKBitRate. by stefan@webrtc.org · 10 years ago
  2. 5647877 Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  3. 0c39e91 Merge beamformer by aluebs@webrtc.org · 10 years ago
  4. 1090a6e Remove obsolete target_arch == armv7. by andrew@webrtc.org · 10 years ago
  5. aacc234 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  6. 16a05dd Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. by jiayl@webrtc.org · 10 years ago
  7. f5847d7 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. by pthatcher@webrtc.org · 10 years ago
  8. cb79141 Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. by asapersson@webrtc.org · 10 years ago
  9. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  10. 98c04b3 Get avg_delay_ms from DecoderTiming callback. by pbos@webrtc.org · 10 years ago
  11. 9b79197 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. by sprang@webrtc.org · 10 years ago
  12. f832a6d Remove _t from function pointer typedefs. by pbos@webrtc.org · 10 years ago
  13. eed7a22 Make an AudioEncoder subclass for iSAC redundant encoding by henrik.lundin@webrtc.org · 10 years ago
  14. dd8f6f3 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. by pbos@webrtc.org · 10 years ago
  15. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  16. e468bc9 Rename _t struct types in audio_processing. by pbos@webrtc.org · 10 years ago
  17. cab1291 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder by henrik.lundin@webrtc.org · 10 years ago
  18. 4fba293 Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port by guoweis@webrtc.org · 10 years ago
  19. 4cb3856 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  20. 536f999 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  21. c51fb93 Fix an assert failure caused by race condition by guoweis@webrtc.org · 10 years ago
  22. 0ab42bc Make safe_conversions suitable for rtc_base_approved. by andrew@webrtc.org · 10 years ago
  23. bc03192 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  24. 0eb6eec Move VirtualSocket into the .h file to allow unit tests more control over behavior. by guoweis@webrtc.org · 10 years ago
  25. 6f10ae2 Support block_size greater than chunk_size in Blocker by aluebs@webrtc.org · 10 years ago
  26. eb54446 Rename _t struct types in audio_coding. by pbos@webrtc.org · 10 years ago
  27. 209df9b Change MockStatsObserver to grab values inside of OnComplete. by tommi@webrtc.org · 10 years ago
  28. e728ee0 Remove or rename typedefs with _t prefixes. by pbos@webrtc.org · 10 years ago
  29. 5263c58 Add a little utility to capture cpu graphs. by tommi@webrtc.org · 10 years ago
  30. 70f74f3 Add overshoot of target bitrate for screenshare with temporal layers. by sprang@webrtc.org · 10 years ago
  31. 45a272a Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls. by asapersson@webrtc.org · 10 years ago
  32. e102e81 Enable the iSACfix AudioDecoder test (and make it work again) by kwiberg@webrtc.org · 10 years ago
  33. 38881be If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport(). by braveyao@webrtc.org · 10 years ago
  34. 950c518 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  35. 971bf55 Fix path to mock_agc.h by andrew@webrtc.org · 10 years ago
  36. f050791 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  37. 4afb599 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  38. e2b7585 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  39. a32487f Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder by henrik.lundin@webrtc.org · 10 years ago
  40. 02c21db Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory. by pthatcher@webrtc.org · 10 years ago
  41. 08df9b2 Add a manageable command-line tool for AudioProcessing. by andrew@webrtc.org · 10 years ago
  42. cf6d0b6 Add 48kHz support to AGC by aluebs@webrtc.org · 10 years ago
  43. 2510d11 Add (safe) uint32_t cast to fix Win64 build. by andrew@webrtc.org · 10 years ago
  44. 048c502 Handle all permissible PCM fields with WavReader. by andrew@webrtc.org · 10 years ago
  45. 451a133 Add AGC manager tests. by pbos@webrtc.org · 10 years ago
  46. c1c9291 Make an AudioEncoder subclass for RED by henrik.lundin@webrtc.org · 10 years ago
  47. 88bdec8 AudioEncoder subclass for iSACfix by kwiberg@webrtc.org · 10 years ago
  48. 0198933 Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate(). by kjellander@webrtc.org · 10 years ago
  49. d08d389 Add field to counters for when first rtp/rtcp packet is sent/received. by asapersson@webrtc.org · 10 years ago
  50. b395a5e audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/ by bjornv@webrtc.org · 10 years ago
  51. 55360ae Revert "Add adapter_type into Candidate object." by guoweis@webrtc.org · 10 years ago
  52. d021bbb Fix vp9 setting in vie loopback test. by marpan@webrtc.org · 10 years ago
  53. aaf02cc Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  54. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  55. 96a6262 Remove 20ms support in AGC by aluebs@webrtc.org · 10 years ago
  56. 1f05c45 Reenable test case P2PTransportChannelTest.TestIPv6Connections by guoweis@webrtc.org · 10 years ago
  57. a7f7772 Merge in AGC manager and AGC tools. by pbos@webrtc.org · 10 years ago
  58. 903b4ae Removes unused test files by audio_processing/transient by bjornv@webrtc.org · 10 years ago
  59. dd32213 resources/audio_processing: Removed unused test files by bjornv@webrtc.org · 10 years ago
  60. 6fd9308 Suppressing warnings in GetRTT() in VoE. by minyue@webrtc.org · 10 years ago
  61. e2e199b Clean up StatsObserver's OnComplete methods (address TODOs). by tommi@webrtc.org · 10 years ago
  62. 3440fe1 Use webrtc_root instead of DEPTH for iSAC. by pbos@webrtc.org · 10 years ago
  63. 032b802 (Auto)update libjingle 82121498-> 82126219 by buildbot@webrtc.org · 10 years ago
  64. dd0601f Remove unneeded ctor and add a more practical one by tommi@webrtc.org · 10 years ago
  65. 69bc5a3 Add thread asserts to StatsCollector. by tommi@webrtc.org · 10 years ago
  66. 788acd1 Merge audio_processing changes. by pbos@webrtc.org · 10 years ago
  67. fb108b5 Revert r7885. by pbos@webrtc.org · 10 years ago
  68. b413a30 Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version. by andrew@webrtc.org · 10 years ago
  69. 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
  70. 40e4767 Add NEON intrinsics version for min_max_operations_neon.c by andrew@webrtc.org · 10 years ago
  71. e575e9c Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h by magjed@webrtc.org · 10 years ago
  72. e9db7fe Put pseudotcp back because remoting uses it. by pthatcher@webrtc.org · 10 years ago
  73. dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  74. 8c9d79a Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  75. c57310b Switch kStatsValueName* constants to be enums instead of char*. by tommi@webrtc.org · 10 years ago
  76. 3b79daf Moving encoded_bytes into EncodedInfo by henrik.lundin@webrtc.org · 10 years ago
  77. c8bc717 Fix webrtc gn windows build. by kjellander@webrtc.org · 10 years ago
  78. f68faa5 Removing manual test pages because they have been moved to github. by jansson@webrtc.org · 10 years ago
  79. 40b276e Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
  80. 27d106b Move the downmixing out of AudioBuffer by aluebs@webrtc.org · 10 years ago
  81. 0ca768b Adding DTX to WebRTC Opus wrapper (relanding). by minyue@webrtc.org · 10 years ago
  82. 5f162c8 Merge AEC changes. by pbos@webrtc.org · 10 years ago
  83. 2b19f06 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
  84. 1351895 Remove old_factory from WebRtcVideoEngine. by pbos@webrtc.org · 10 years ago
  85. 128faba Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" by perkj@webrtc.org · 10 years ago
  86. 626c09f Move isolate path into webrtc/build/android/test_runner.py by kjellander@webrtc.org · 10 years ago
  87. 817e50d Make an AudioEncoder subclass for PCM16B by henrik.lundin@webrtc.org · 10 years ago
  88. b3ad8cf Make an AudioEncoder subclass for iSAC by kwiberg@webrtc.org · 10 years ago
  89. abe3f18 Checking whether ACM uses codec internal or WebRTC DTX. by minyue@webrtc.org · 10 years ago
  90. 55d42c3 DCHECK: Reference condition parameter in release builds by kwiberg@webrtc.org · 10 years ago
  91. cd5b209 Deleting quality dashboard code. by phoglund@webrtc.org · 10 years ago
  92. 3c31e6e Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon by andrew@webrtc.org · 10 years ago
  93. f4c1948 Remove jitter_estimate_test.h by mflodman@webrtc.org · 10 years ago
  94. c5ebbd9 Support 48kHz in Noise Suppression by aluebs@webrtc.org · 10 years ago
  95. d8ca723 Remove CELT support from audio_coding. by pbos@webrtc.org · 10 years ago
  96. 8084f95 Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. by asapersson@webrtc.org · 10 years ago
  97. 85bd53e Add AbsSendTime unittests to rampup_tests.cc. by pbos@webrtc.org · 10 years ago
  98. 0df3715 Cast payload type to int in logs. by asapersson@webrtc.org · 10 years ago
  99. a853077 (Auto)update libjingle 81702493-> 81755413 by buildbot@webrtc.org · 10 years ago
  100. 3cd26b6 Revert r7858 ("DCHECK: Reference condition parameter in release builds") by kwiberg@webrtc.org · 10 years ago