Google Git
Sign in
webrtc / src / HEAD
  1. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  2. 2c850af Use nalus_length instead of is_first_packet_in_frame to insert startcodes in the H264SpsPpsTracker. by philipel · 5 years ago
  3. 6564366 Use -Wno-undef to compile jsoncpp. by Mirko Bonadei · 5 years ago
  4. d404946 Minor modifications for ADM2 on Windows. by henrika · 5 years ago
  5. a0e2609 Partially revert of ColorSpace information copying around decoders by Ilya Nikolaevskiy · 5 years ago
  6. 8b3e4e2 Revert "Reland "Add ability to set RTCP sender ssrc at construction time"" by Mirko Bonadei · 5 years ago
  7. 01d04fa Revert "Add wrapper to normalize flags." by Artem Titarenko · 5 years ago
  8. 0bb0881 Add VideoEncoderFactory::GetImplementations function. by philipel · 5 years ago
  9. 66147e8 Revert "Optimize PacketRouter/RTPSender interactions." by Mirko Bonadei · 5 years ago
  10. 4d68314 Revert "Pass RtpRtcp::Configuration to RtcpReceiver ctor and initialize ssrcs" by Mirko Bonadei · 5 years ago
  11. 1182050 Revert "Make new pacer padding more like old one" by Mirko Bonadei · 5 years ago
  12. 32b1834 Use default task queue factory in fuzzers by Danil Chapovalov · 5 years ago
  13. 44cec0b Handle non-integer frame rates in video codec tests. by Sergey Silkin · 5 years ago
  14. 3ae59d3 Use the dummy address 0.0.0.0:9 in the c= and the m= lines if the by Qingsi Wang · 5 years ago
  15. 58388aa Roll chromium_revision 3bcb109517..ade23986de (676409:676659) by chromium-webrtc-autoroll · 5 years ago
  16. 6ff9ebd Revert "Refactor FEC code to use COW buffers" by Qingsi Wang · 5 years ago
  17. 0f0668e Revert "Cleanup FEC code after refactoring" by Qingsi Wang · 5 years ago
  18. 66b3860 Remove WebRTC-SimulcastScreenshare and enable it by default by Florent Castelli · 5 years ago
  19. ac6c096 Integrate datagram feedback loop by Anton Sukhanov · 5 years ago
  20. 1e19db1 Let git-hyper-blame ignore new format cleanup. by Yves Gerey · 5 years ago
  21. 4e5a41a Cleanup FEC code after refactoring by Ilya Nikolaevskiy · 5 years ago
  22. 7325bc3 Refactor FEC code to use COW buffers by Ilya Nikolaevskiy · 5 years ago
  23. 1300ce1 Roll chromium_revision 8bdf10dfd2..3bcb109517 (676126:676409) by chromium-webrtc-autoroll · 5 years ago
  24. bb77272 Make new pacer padding more like old one by Erik Språng · 5 years ago
  25. 741b96b Pass RtpRtcp::Configuration to RtcpReceiver ctor and initialize ssrcs by Erik Språng · 5 years ago
  26. 6f129b3 Optimize PacketRouter/RTPSender interactions. by Erik Språng · 5 years ago
  27. 08511d2 Roll chromium_revision fff68b4152..8bdf10dfd2 (675999:676126) by chromium-webrtc-autoroll · 5 years ago
  28. ee303fa Move datagram_dtls_adaptor from p2p/base/ to pc/ by Anton Sukhanov · 5 years ago
  29. 642a49d Add wrapper to normalize flags. by Mirko Bonadei · 5 years ago
  30. 52e240e Use 16000Hz audio in PC test when specified by Artem Titov · 5 years ago
  31. ff25b87 Implements method on RtpPacket to extract extension. by Anton Sukhanov · 5 years ago
  32. ca5f21e Make force_fieldtrials persistent string during entire program live. by Ruslan Burakov · 5 years ago
  33. a1471b7 Roll chromium_revision 34d38f69ba..fff68b4152 (675873:675999) by chromium-webrtc-autoroll · 5 years ago
  34. b1f2d60 Reland "Fix collection of audio metrics from PC test framework for audio test" by Artem Titov · 5 years ago
  35. ca16021 Update rtc_tools/rtp_generator to compile by Ilya Nikolaevskiy · 5 years ago
  36. 1685059 Add support of quick test mode into PC framework by Artem Titov · 5 years ago
  37. 41300af Poison default task queue factory by Danil Chapovalov · 5 years ago
  38. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  39. 0d90a1d Do not use hungarian notation for DwmGetWindowAttribute's params by Julien Isorce · 5 years ago
  40. d6f4f74 Roll chromium_revision 58be81bf4b..34d38f69ba (675720:675873) by chromium-webrtc-autoroll · 5 years ago
  41. 6f420e4 Reland "Add ability to set RTCP sender ssrc at construction time" by Erik Språng · 5 years ago
  42. 4876cb2 Revert "Fix collection of audio metrics from PC test framework for audio test" by Mirko Bonadei · 5 years ago
  43. 49fa4ea Roll chromium_revision 2bc3837c3d..58be81bf4b (675350:675720) by chromium-webrtc-autoroll · 5 years ago
  44. d0679bd Enables usage of ChannelMixer in WebRTC's output mixer. by henrika · 5 years ago
  45. b249c54 Delete GlobalTaskQueueFactory as now unused by Danil Chapovalov · 5 years ago
  46. 6456e35 Use max bitrate limit recommended by encoder. by Sergey Silkin · 5 years ago
  47. 2d0880b Fix collection of audio metrics from PC test framework for audio test by Artem Titov · 5 years ago
  48. bc558ce Add support of specifying audio sample rate for PC test framework by Artem Titov · 5 years ago
  49. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 5 years ago
  50. cb60a8b Roll chromium_revision b624ecb939..2bc3837c3d (675206:675350) by chromium-webrtc-autoroll · 5 years ago
  51. 5e25fac CroppingWindowCapturerWin: filter out cloaked window. by Julien Isorce · 5 years ago
  52. a0eefc1 Rename USE_NATIVE_MUTEX_ON_MAC to RTC_USE_NATIVE_MUTEX_ON_MAC. by Mirko Bonadei · 5 years ago
  53. 34462f5 Revert "Add ability to set RTCP sender ssrc at construction time" by Ilya Nikolaevskiy · 5 years ago
  54. 4a126e4 Rename tests to prevent clashing with old audio test by Artem Titov · 5 years ago
  55. 94c58fd Add ability to set RTCP sender ssrc at construction time by Erik Språng · 5 years ago
  56. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  57. c93bfcf Roll chromium_revision 619b1dc518..b624ecb939 (675093:675206) by chromium-webrtc-autoroll · 5 years ago
  58. 48284b8 BalancedDegradationSettings: Add option to configure fps based on codec type. by Åsa Persson · 5 years ago
  59. 79e4c92 Remove bwe_rtp_play and add rtp_to_text to the build. by Mirko Bonadei · 5 years ago
  60. ab0b9d4 Switch rtc_event_log2rtp_dump to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  61. eec86cd Fix platform-specific header dependencies to be more precise by Oleh Prypin · 5 years ago
  62. 86f8b3b Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093) by chromium-webrtc-autoroll · 5 years ago
  63. 9455c30 Remove deprecation notice for VideoFrame::ntp_time_ms by Ilya Nikolaevskiy · 5 years ago
  64. f6468d2 Wire up new PacedSender code path. by Erik Språng · 5 years ago
  65. 668ce0c Remove trial WebRTC-SimulcastMaxLayers and make its behavior default by Florent Castelli · 5 years ago
  66. ecae9cd Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer by Magnus Jedvert · 5 years ago
  67. 48b1b18 Add ability to create EmulatedNetworkNode from BuiltInNetworkBehaviorConfig by Artem Titov · 5 years ago
  68. c8263e0 Introduce PC level audio quality test. by Artem Titov · 5 years ago
  69. 386802e Move network emulation framework under test/network by Artem Titov · 5 years ago
  70. 13eb602 Roll chromium_revision 50acc956cd..48f88caf2e (674882:674992) by chromium-webrtc-autoroll · 5 years ago
  71. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 5 years ago
  72. 656590d Roll chromium_revision 778353d874..50acc956cd (674780:674882) by chromium-webrtc-autoroll · 5 years ago
  73. 1efb4a2 Add field trial for forcing partition resilience mode in libvpx. by Rasmus Brandt · 5 years ago
  74. 3fbf1e2 Reduce kMaxSimulcastStreams to 3 by Florent Castelli · 5 years ago
  75. 9d96209 Switch unpack_aecdump to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  76. 4580ca2 Reland: Add ability to set ssrcs of RtpSender at construction time by Erik Språng · 5 years ago
  77. 8e3cb53 Remove activity_metric tool. by Mirko Bonadei · 5 years ago
  78. 543179e Roll chromium_revision 79b588ee95..778353d874 (674667:674780) by chromium-webrtc-autoroll · 5 years ago
  79. 2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
  80. 3f2eeb8 Adding test on GetSpanSamples() for NetEq PacketBuffer. by Minyue Li · 5 years ago
  81. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  82. e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
  83. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  84. c6c730b Roll chromium_revision 54ec0ffb89..79b588ee95 (674538:674667) by chromium-webrtc-autoroll · 5 years ago
  85. 697f861 Roll chromium_revision 13b7238371..54ec0ffb89 (674397:674538) by chromium-webrtc-autoroll · 5 years ago
  86. 2e60217 Add speculative checks to RtpPacketHistory by Erik Språng · 5 years ago
  87. 46dda83 Improve buffer level estimation with DTX and add CNG time stretching. by Jakob Ivarsson · 5 years ago
  88. 3d642f8 Rename ..BitrateThresholds to ..BitrateLimits. by Sergey Silkin · 5 years ago
  89. cecee99 Disable VP9 2nd profile test for ios arm64 by Artem Titov · 5 years ago
  90. 51f599b Make rtc_base/base_java public. by Sami Kalliomäki · 5 years ago
  91. 81e1bf0 Remove using DegradationPreference from scenario_config.h by Artem Titov · 5 years ago
  92. 6542826 Add new tests with lossy networks on PC test framework by Artem Titov · 5 years ago
  93. cd8a6e2 Add writing and parsing of the `abs-capture-time` RTP header extension. by Chen Xing · 5 years ago
  94. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  95. 84ce3c0 Macro rename s/CS_DEBUG_CHECKS/RTC_CS_DEBUG_CHECKS. by Mirko Bonadei · 5 years ago
  96. a2b30d8 Add functions to read from/write to bitstream values with known max value by Danil Chapovalov · 5 years ago
  97. 9eee121 Switch py_quality_assessment to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  98. b60141b Save and serialize the receive RIDs in MediaContentDescription by Florent Castelli · 5 years ago
  99. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 5 years ago
  100. e420c6a Add missing include for memcpy/memcmp by Artem Titov · 5 years ago