1. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  2. 92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 7 years ago
  3. 247e0b4 Disabling periodic tasks on SSCC in unit tests. by Sebastian Jansson · 7 years ago
  4. 81e8a43 Roll chromium_revision 3ad75d0441..89bed79700 (544573:544673) by Autoroller · 7 years ago
  5. 29b204e Tracking packet feedback availability in BitrateAllocator. by Sebastian Jansson · 7 years ago
  6. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  7. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  8. c3d1e09 Make sure RTCMTLVideoView.h ends up in framework headers. by Anders Carlsson · 7 years ago
  9. 08006d4 Android AppRTCMobile: Use new audio device code by Magnus Jedvert · 7 years ago
  10. 82fad3d Remove TemporalLayersFactory and associated classes by Erik Språng · 7 years ago
  11. 8fc7948 Android: Generate audio JNI code by Magnus Jedvert · 7 years ago
  12. 37e3602 Android: Add henrika@ as owner of audio code by Magnus Jedvert · 7 years ago
  13. e7fac68 Introduce Nullable annotation. by Sami Kalliomäki · 7 years ago
  14. 7531a76 Delete unused header media/base/test/mock_mediachannel.h. by Niels Möller · 7 years ago
  15. 9611442 Remove unneeded migration helper. by Sami Kalliomäki · 7 years ago
  16. e61631d Roll chromium_revision 2f0e6b63b5..3ad75d0441 (544446:544573) by Autoroller · 7 years ago
  17. b88bfc9 Roll chromium_revision 8c0344a12e..2f0e6b63b5 (544337:544446) by Autoroller · 7 years ago
  18. db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
  19. e9d2e4d Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods. by Dino Radaković · 7 years ago
  20. fe48ee9 Fixing zlib license generation. by Mirko Bonadei · 7 years ago
  21. b38b05b Adding srte as owner in modules/congestion_controller. by Sebastian Jansson · 7 years ago
  22. 89dd7bf Move android audio device code into sdk/android by Paulina Hensman · 7 years ago
  23. 4d22a6d Delete unneeded includes of wav_file.h and file_wrapper.h. by Niels Möller · 7 years ago
  24. 8ef59a4 Added data member access methods to FakeNetworkPipe. by Christoffer Rodbro · 7 years ago
  25. 3dc0125 Moving ConfigureEncoderTask to the calling scope. by Sebastian Jansson · 7 years ago
  26. 68a7168 Roll chromium_revision ee966518c2..8c0344a12e (544233:544337) by Autoroller · 7 years ago
  27. 2e0da5a Remove EncodedFrame picture_id/spatial_layer references. by philipel · 7 years ago
  28. f18072e Enable SVC based on number of SSRCs. by Sergey Silkin · 7 years ago
  29. eb98c72 Minor improvements in ADM unittest for Windows. by henrika · 7 years ago
  30. faed538 Delete obsolete alias RateLimiter and rtc_base/ratelimiter.h by Niels Möller · 7 years ago
  31. 9047dac Disable flaky test SendSideCongestionControllerTest/PacerQueueEncodeRatePushback. by Rasmus Brandt · 7 years ago
  32. d328ec3 Change valueOf -> parseBoolean to avoid unneeded boxing by Oleh Prypin · 7 years ago
  33. a98bb2d Roll chromium_revision 02de71e34e..ee966518c2 (544129:544233) by Autoroller · 7 years ago
  34. 114155b Roll chromium_revision 163641576c..02de71e34e (544029:544129) by Autoroller · 7 years ago
  35. 01cb5f2 Fix issue where sockets bound to temporary IPv6 addresses are discarded. by Taylor Brandstetter · 7 years ago
  36. 3d976f6 Discard link to media channel when audio sender stopped. by Harald Alvestrand · 7 years ago
  37. bb60a3a Refactor VP8 TemporalLayers by Erik Språng · 7 years ago
  38. d757356 Fixing -Wstrict-prototypes warnings. by Mirko Bonadei · 7 years ago
  39. def1ef5 New equality operators, for structs related to webrtc::VideoCodec. by Niels Möller · 7 years ago
  40. 5b3541f RTCStatsCollector::GetStatsReport() with optional selector argument. by Henrik Boström · 7 years ago
  41. 4a73cd4 Adding tests of TaskQueueCongestionControl field trial. by Sebastian Jansson · 7 years ago
  42. e62f600 Extend WavReader and WavWriter API. by Artem Titov · 7 years ago
  43. 451dfdf Roll chromium_revision 7a8a322ad7..163641576c (543921:544029) by Autoroller · 7 years ago
  44. 0fa82a6 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId. by philipel · 7 years ago
  45. 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 7 years ago
  46. b9a02e5 Change place of UMA logging in AudioMixer. by Alex Loiko · 7 years ago
  47. 5370124 Replacing unique pointer with raw pointer in SSCC checks. by Sebastian Jansson · 7 years ago
  48. 04d4950 Revert "Using safe casts of allocation limits in Call." by Oleh Prypin · 7 years ago
  49. 4a9b4d6 Using safe casts of allocation limits in Call. by Sebastian Jansson · 7 years ago
  50. 8d8cb56 Delete obsolete methods from MockRtpTransportControllerSend by Sebastian Jansson · 7 years ago
  51. b708e93 Bring mb up to date with Chromium's changes by Oleh Prypin · 7 years ago
  52. 8d2c5a8 Detangling target dependencies in rtc_base_approved. by Tommi · 7 years ago
  53. 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 7 years ago
  54. d2c8332 Revert "Relaxing no-streams presubmit check (streams are allowed in tests)." by Mirko Bonadei · 7 years ago
  55. 7311918 Add an example app for iOS native API. by Anders Carlsson · 7 years ago
  56. 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  57. 73ac908 Relaxing no-streams presubmit check (streams are allowed in tests). by Mirko Bonadei · 7 years ago
  58. 9fa35e5 Fix path to proto in py_event_log_analyzer/pb_parse.py by Oleh Prypin · 7 years ago
  59. 8a1b20a Roll chromium_revision 33aa22e76e..7a8a322ad7 (543816:543921) by Autoroller · 7 years ago
  60. cdd2a97 Roll chromium_revision ce851e47bd..33aa22e76e (543685:543816) by Autoroller · 7 years ago
  61. 317a522 Fixes to posting delayed process tasks in SSCC. by Sebastian Jansson · 7 years ago
  62. 4ccc1c4 Don't destroy a receive stream's sink before reassigning it. by Oskar Sundbom · 7 years ago
  63. 3bb1194 Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'" by Patrik Höglund · 7 years ago
  64. 3546835 Roll chromium_revision e0e02de5a7..ce851e47bd (543578:543685) by Autoroller · 7 years ago
  65. bf3dbb4 Delete payload_type from VCMEncoderDatabase and vcm::VideoSender. by Niels Möller · 7 years ago
  66. 5bf8ccd Delete encoder caching in WebRtcVideoSendStream. by Niels Möller · 7 years ago
  67. 677f42c Enable ContinuousAfterStreamCountChangeSimulcastEncoderAdapter picture id tests. by Åsa Persson · 7 years ago
  68. d132ce1 Remove unnecessary copies from AsyncInvoke by Cameron Pickett · 7 years ago
  69. 465a5d9 Refactor payload types constants in CallTest by Ilya Nikolaevskiy · 7 years ago
  70. af9e87b Delete unused methods from vcm::VideoCodingModule. by Niels Möller · 7 years ago
  71. eef09fc Fix race in DegradedCall::DestroyVideoSendStream by Erik Språng · 7 years ago
  72. 883d00f Add support of AAudio in native WebRTC on Android O and above by henrika · 7 years ago
  73. 815f3b6 Fix podspec iOS version. by Kári Tristan Helgason · 7 years ago
  74. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  75. 4de9eb2 Roll chromium_revision bc2c5b551b..e0e02de5a7 (543473:543578) by Autoroller · 7 years ago
  76. 8870f55 Roll chromium_revision d94f7320ab..bc2c5b551b (543368:543473) by Autoroller · 7 years ago
  77. 24c220c Changed target_angle_degrees in audioproc_float to float to avoid integer division when converting to radians by Alex Luebs · 7 years ago
  78. 895ae9a Improving the speed of the delay estimator in AEC3 by Per Åhgren · 7 years ago
  79. 1d037ae Don't crash in SingleNalu packetization for h264 if no space in packet by Ilya Nikolaevskiy · 7 years ago
  80. 4425b05 Add video send stream test to check switch to and from screenshare by Ilya Nikolaevskiy · 7 years ago
  81. aca5a7d Improvements to network control types. by Sebastian Jansson · 7 years ago
  82. dfe6bcd Roll chromium_revision 670c468885..d94f7320ab (543262:543368) by Autoroller · 7 years ago
  83. 56f9f0e Make task_queue_ injectable by wrapping it into a std::unique_ptr and adding an optional arg to the constructor of RtcEventLogImpl. by Dino Radaković · 7 years ago
  84. 3ab1d26 Exposing WebRTC-Audio-SendSideBwe-For-Video field trial by Alex Narest · 7 years ago
  85. 650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 7 years ago
  86. 63b7574 Add check for negative max bitrate in VideoSendStream. by Sebastian Jansson · 7 years ago
  87. 7bd79a0 Split up audio_device build target by Paulina Hensman · 7 years ago
  88. 5f1a31c Adding a smooth transition from the startup phase parameter set in AEC3 by Per Åhgren · 7 years ago
  89. b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  90. 2cb4105 Moved audioproc_f interface into api directory. by Ivo Creusen · 7 years ago
  91. bb894ff Make PayloadRouter own the picture id and tl0 pic idx sequences. by Niels Möller · 7 years ago
  92. 9f64b9c Reland "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
  93. 853715c Min BWE default is 10kbps but for audio send side BWE it was overridden to 5kbps. Now audio send side BWE is used for video calls too and should set min to 10kbps in case of video call. by Alex Narest · 7 years ago
  94. cc681cc Split vp8_impl into webm_vp8_encoder and webm_vp8_decoder by Erik Språng · 7 years ago
  95. b3bac5e Revert "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
  96. 6f2fcb4 Add more Audio Mixer and Fixed Gain Controller metrics. by Alex Loiko · 7 years ago
  97. aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  98. e0eb13c Remove unnecessary dependency on base. by Patrik Höglund · 7 years ago
  99. 180d992 Style guide: State what version of C++ we should use by Karl Wiberg · 7 years ago
  100. 4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 7 years ago