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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 3bd659f1040420b7ca90d0e7df368b083148d0e5 / modules / rtp_rtcp
  1. 64b5c61 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
  2. 7485573 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
  3. b676ac7 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
  4. 4e7777b Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
  5. e97b69f Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
  6. 7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  7. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  8. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  9. 00c95bf Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
  10. 1cc93a2 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 11 years ago
  11. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  12. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  13. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  14. 49bc1b8 Fixes to padding when driven by encoder. by stefan@webrtc.org · 11 years ago
  15. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  16. 03931c6 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 11 years ago
  17. a6b178f Fix duplicate code by niklas.enbom@webrtc.org · 11 years ago
  18. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  19. 9d71e28 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  20. 1c8d5a0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  21. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  22. d0cfc70 Make sure first RTP packet counts as in-order. by pbos@webrtc.org · 11 years ago
  23. cb0c159 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" by elham@webrtc.org · 11 years ago
  24. 9d788a1 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" by elham@webrtc.org · 11 years ago
  25. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  26. 4981d3c Revert r4328 by elham@webrtc.org · 11 years ago
  27. 7b66e14 Fixes a crash when sending SR reports from a sender only module. by stefan@webrtc.org · 11 years ago
  28. b48a4a9 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. by braveyao@webrtc.org · 11 years ago
  29. d3756f7 Sorted headers under rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  30. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  31. 446ea2e Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 11 years ago
  32. e37c2cd Fix three uninitialized members in rtp_receiver_impl.cc. by stefan@webrtc.org · 11 years ago
  33. d5e5863 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  34. f6d9630 Create gyp target for bwe components. by stefan@webrtc.org · 11 years ago
  35. 9c0f14d Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
  36. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  37. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  38. 0be9202 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. by solenberg@webrtc.org · 11 years ago
  39. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  40. 8ee45da Make sure padding packets are sent. by stefan@webrtc.org · 11 years ago
  41. 0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  42. 7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  43. 69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  44. 8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
  45. 789e98b Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
  46. f04f54a Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
  47. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  48. 12bce3b Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
  49. a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  50. cbd78ae Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  51. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  52. 2dcf742 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
  53. 294b789 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
  54. 68c4886 Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
  55. aef3e5a Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
  56. eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  57. 8f1d1a9 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
  58. 0e15695 Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. by solenberg@webrtc.org · 11 years ago
  59. 4efbd60 Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
  60. a0b0025 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
  61. b960975 Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
  62. 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  63. efbf737 Reformatted FEC tables. by phoglund@webrtc.org · 11 years ago
  64. d474c13 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
  65. a149ea3 Formatted dtmf_queue. by phoglund@webrtc.org · 11 years ago
  66. 98b2011 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
  67. 52b2ee5 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
  68. 36bdba4 Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
  69. 2d6f0df Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
  70. e422d12 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
  71. b8f1cf3 Enable Nack pacing. by pwestin@webrtc.org · 11 years ago
  72. d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  73. 2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  74. 237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  75. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  76. 9b53152 Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  77. 42c7409 Fix issues with incorrect wrap checks when having big buffers and high bitrate. by stefan@webrtc.org · 11 years ago
  78. bda02e4 Re-write the build of the nacklist. by andresp@webrtc.org · 11 years ago
  79. 0486a10 Removing remaining WebRtc_Word32 not in typedefs.h by pbos@webrtc.org · 11 years ago
  80. 74472fe More trace events by hclam@chromium.org · 11 years ago
  81. 73ebe67 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
  82. 65deb26 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
  83. b57da65 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  84. 1d25eac Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  85. 2ed1cd9 Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 11 years ago
  86. dded206 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
  87. 88f12ab Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
  88. dca71b2 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  89. aa0fcd7 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
  90. e3339fc Generic video-codec support. by pbos@webrtc.org · 11 years ago
  91. 0b5c7f1 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
  92. 946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  93. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  94. f0f1dc2 Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
  95. 3c365b2 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 11 years ago
  96. 15fc445 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 11 years ago
  97. 46e08da Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 11 years ago
  98. d1932fc Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. by bemasc@google.com · 11 years ago
  99. 6e7945f Fix to send a full NACK list at least roughly once every 1.5 x RTT. by stefan@webrtc.org · 12 years ago
  100. 55e6f58 Stop and restart fix. by mflodman@webrtc.org · 12 years ago