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gerrit-public.fairphone.software / platform / external / webrtc / 13d38fbe904cd99acebf77068e93df34725eeb47
  1. 13d38fb Delete all of the video_processing module but the denoiser code. by nisse · 8 years ago
  2. 13f1a0a Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel. by stefan · 8 years ago
  3. 127387e Delete nalu parser in mediaencoder by kthelgason · 8 years ago
  4. 2305b70 Remove unused build override by kthelgason · 8 years ago
  5. cb861e0 Templatize percentile_filter.h and move it to base/analytics. by terelius · 8 years ago
  6. 4b9a2cb Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate." by minyue · 8 years ago
  7. 26bddb9 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac by ehmaldonado · 8 years ago
  8. 18f7c8d Remove warning suppression from VideoToolboxEncoder by kthelgason · 8 years ago
  9. 076c011 Change unit of logged bitrate stats in bytes/s to bits/s. by asapersson · 8 years ago
  10. aff9636 Greatly reduce number of level controller tests. by phoglund · 8 years ago
  11. 6a2e20a Make sure GetLastError on a PlatformThread return an error that is relevant to the thread. by perkj · 8 years ago
  12. b49fc14 RtpDataEngine, FindCodecByName: Don't reassign codecs by magjed · 8 years ago
  13. 998df1d Create webrtc/sdk/android folder by magjed · 8 years ago
  14. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  15. 8d66a5a Disabled flaky P2PTestConductor tests on ASAN and MSAN. by ossu · 8 years ago
  16. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  17. 06251f1 Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps. by philipel · 8 years ago
  18. f4a5942 Disabled all ScreenCapturerIntegrationTests on Windows by ossu · 8 years ago
  19. 3a864d2 MB: Add swarming bots in the FYI waterfall and remove memcheck swarming. by ehmaldonado · 8 years ago
  20. b7e1dd7 Revert of Adding memcheck suppression. (patchset #1 id:1 of https://codereview.webrtc.org/2537563003/ ) by philipel · 8 years ago
  21. 706a45e Added missing include to fix waterfall compile error. by hbos · 8 years ago
  22. f15a2c5 Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom. by nisse · 8 years ago
  23. 0583b28 Collecting RTCIceCandidatePairStats.transport_id and improved unittests. by hbos · 8 years ago
  24. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  25. 759e0b7 Fix memory leak in video_coding::PacketBuffer::InsertPacket. by philipel · 8 years ago
  26. be74270 Calculate JitterBufferDelayInMs in the new jitter buffer. by philipel · 8 years ago
  27. e69b468 Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ ) by minyue · 8 years ago
  28. 1731c9c Use swap instead of copy in RtcHistogram::GetAndReset. by asapersson · 8 years ago
  29. 84e56d5 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. by minyue · 8 years ago
  30. 097529f Remove 3 defines in voice_engine_configurations.h by henrik.lundin · 8 years ago
  31. e61fbff Use RotateDesktopFrame in DirectX capturer by zijiehe · 8 years ago
  32. 166e59a Enable ScreenCapturerIntegrationTests by zijiehe · 8 years ago
  33. 6a46cf7 Roll chromium_revision db14e1adbd..b66d8ae9dc (435041:435081) by buildbot · 8 years ago
  34. c9e80ee Adding packet overhead to audio network adaptor. by minyue · 8 years ago
  35. 821dc7a Roll chromium_revision 683745f53c..db14e1adbd (434997:435041) by buildbot · 8 years ago
  36. a332877 Remove overhead from video bitrate. by michaelt · 8 years ago
  37. c4dc4a5 Adding RTCStatsIntegrationTest to memcheck supressions. by deadbeef · 8 years ago
  38. 75f9d8c Roll chromium_revision ffe8e7b51d..683745f53c (434954:434997) by buildbot · 8 years ago
  39. 290d43a Add a new UMA metric in APM to track incoming capture-side audio level by henrik.lundin · 8 years ago
  40. 939e08f Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
  41. ee414d9 Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error. by philipel · 8 years ago
  42. ad6f646 Use //build/dotfile_settings.gni to reduce blocked auto-rolls by kjellander · 8 years ago
  43. 768d625 Fix spelling mistake in RTP module declaration. by brandtr · 8 years ago
  44. b890c95c Replace some asserts with DCHECKs by kwiberg · 8 years ago
  45. 5049942 Refactor RMSLevel and give it new functionality by henrik.lundin · 8 years ago
  46. 1308c69 Roll chromium_revision 0496be2799..ffe8e7b51d (434847:434954) by buildbot · 8 years ago
  47. f17cae2 Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it by kjellander · 8 years ago
  48. 668eb3b Add overhead to transport feedback observer. by michaelt · 8 years ago
  49. 19223ac Ignore newly added resource files. by charujain · 8 years ago
  50. 455b512 Landmine to clobber Windows builders by Henrik Kjellander · 8 years ago
  51. 1b5b22d Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  52. db346a7 RTCStatsIntegrationTest added. by hbos · 8 years ago
  53. 876222f Move usage of QualityScaler to ViEEncoder. by kthelgason · 8 years ago
  54. 320e45a Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  55. 65e5f5a Roll chromium_revision d74a300097..0496be2799 (434704:434847) by buildbot · 8 years ago
  56. bdd6f4c Adding memcheck suppression. by deadbeef · 8 years ago
  57. 6cf94a0 Only use BoringSSL time callback in unit tests. by deadbeef · 8 years ago
  58. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  59. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  60. 80ed35e Implement periodic bandwidth probing in application-limited region. by sergeyu · 8 years ago
  61. bf22be9 Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704) by buildbot · 8 years ago
  62. fd87f4a Opus: Move complexity variable out of conditional build flag by henrik.lundin · 8 years ago
  63. 1bc3146 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit by Henrik Kjellander · 8 years ago
  64. bb58435 Fix potential synchronization issues with framelisteners in EglRenderer. by sakal · 8 years ago
  65. 266f0a4 Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment. by philipel · 8 years ago
  66. d1aaaaa Remove surface size mismatch logic from EglRenderer. by sakal · 8 years ago
  67. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  68. 7703b27 Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots. by philipel · 8 years ago
  69. ceecea4 Pass selected cricket::VideoCodec down to internal H264 encoder by magjed · 8 years ago
  70. 20dce34 Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called. by philipel · 8 years ago
  71. e1a13f8 MB: Remove a --target-devices-file flag for JUnit tests on android. by ehmaldonado · 8 years ago
  72. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  73. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  74. ffc6118 Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
  75. ec1a670 Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream. by Rasmus Brandt · 8 years ago
  76. fb4a37a Add memcheck suppressions. by ehmaldonado · 8 years ago
  77. 26fa6b2 Revert of Bug in ExtractFrame API (extracts frames incorrectly) (patchset #9 id:130001 of https://codereview.webrtc.org/2529923002/ ) by charujain · 8 years ago
  78. 566cba1 Roll chromium_revision 5c22c2afac..2b5aa49038 (434448:434640) by buildbot · 8 years ago
  79. b7636b4 Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  80. 2f58ec8 Add I420Buffer::Copy method taking plane pointers as input. by nisse · 8 years ago
  81. e441bdb Cleanup RtpSender hiding RtpHeaderExtensionLength function. by danilchap · 8 years ago
  82. 2fedf9c Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
  83. 847f689 Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448) by kjellander · 8 years ago
  84. deb95f3 Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t. by nisse · 8 years ago
  85. 71b9b58 Revert of Move ADM specific Android files into modules/audio_device/android/ (patchset #2 id:20001 of https://codereview.webrtc.org/2533573002/ ) by solenberg · 8 years ago
  86. e8d8a2b Move ADM specific Android files into modules/audio_device/android/ by solenberg · 8 years ago
  87. e69a1a9 Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ ) by magjed · 8 years ago
  88. d7e6ccb Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ ) by magjed · 8 years ago
  89. c7805db Fix perf regression in screenshare temporal layer bitrate allocation by sprang · 8 years ago
  90. fd34d30 iOS HW encoder: Enable H264 High profile support by magjed · 8 years ago
  91. bdbc4b7 Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload by magjed · 8 years ago
  92. 1da1a09 Android HW encoder: Set constrained baseline as the profile by magjed · 8 years ago
  93. 03d6b08 Get rid of webrtc/base/latebindingsymboltable* by ehmaldonado · 8 years ago
  94. f3feeff Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  95. 0fa164a Make Valgrind memcheck work in swarming. by ehmaldonado · 8 years ago
  96. 5732910 Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ ) by ehmaldonado · 8 years ago
  97. 76622ce Adding a unit test for RMSLevel by henrik.lundin · 8 years ago
  98. 293bc2a Add 'Update LASTCHANGE' hook to DEPS by ehmaldonado · 8 years ago
  99. 5f7226f Turn off error resilience for vp8 for no temporal layers if nack is enabled. by asapersson · 8 years ago
  100. 5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago