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android-review.linaro.org / platform / external / chromium_org / third_party / webrtc / eb46bb84a37bc9384ec3941a97966bec87d2476c
  1. eb46bb8 Volume buttons in AppRTCDemo should affect output audio volume (part II). by henrika@webrtc.org · 10 years ago
  2. 98dd0b8 Log formatting fix for VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  3. 3ebce78 Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  4. b831a9e Adjust parameter in vp9 rate control test. by marpan@webrtc.org · 10 years ago
  5. 10d91f4 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. by marpan@webrtc.org · 10 years ago
  6. 941173a Remove uses of build date/time. by pbos@webrtc.org · 10 years ago
  7. 5232267 Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  8. 048258f Restore old behavior for Android in fileutils.cc by kjellander@webrtc.org · 10 years ago
  9. 20b2dc6 Fix android_clang build. by glaznev@webrtc.org · 10 years ago
  10. d1f71cc Revert 7623 "Remove the state_ member from AudioDecoder" by niklas.enbom@webrtc.org · 10 years ago
  11. 1145210 Revert 7625 "Don't use DCHECK when you need the side effects..." by niklas.enbom@webrtc.org · 10 years ago
  12. ff6cd0b Don't use DCHECK when you need the side effects... by kwiberg@webrtc.org · 10 years ago
  13. 5721e27 Remove the state_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  14. 385606f Add support for VP9 in webrtc::Call and video_loopback. by stefan@webrtc.org · 10 years ago
  15. c4cd81a Reduce to 2 probes when probing for initial bandwidth. by stefan@webrtc.org · 10 years ago
  16. 4706c96 Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged. by stefan@webrtc.org · 10 years ago
  17. 9a8c28f Reworked paced sender queue by sprang@webrtc.org · 10 years ago
  18. 8d28158 Adds support for finch experiments to video_loopback. by stefan@webrtc.org · 10 years ago
  19. 95e0f61 Fix problem with late packets in NetEq by henrik.lundin@webrtc.org · 10 years ago
  20. e765eef Delete VideoReceiveStream channels in destructor. by pbos@webrtc.org · 10 years ago
  21. ab22837 Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 by kwiberg@webrtc.org · 10 years ago
  22. 14f28eb Remove the useless dummy state parameter to WebRtcG711_* by kwiberg@webrtc.org · 10 years ago
  23. a1fd19c Remove the codec_type_ member from AudioDecoder by kwiberg@webrtc.org · 10 years ago
  24. 343d3cf Enables AIMD control by default. by stefan@webrtc.org · 10 years ago
  25. dba94e1 Improving error message from neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  26. deb9e49 Add format members to AudioConverter for DCHECKing. by andrew@webrtc.org · 10 years ago
  27. afcd610 Update rate control parameter in vp9 test. by marpan@webrtc.org · 10 years ago
  28. dd298d8 Roll chromium_revision: 28d1981..d3db2ff by marpan@webrtc.org · 10 years ago
  29. 1c755a6 Restore the void return type on WriteWavHeader. by andrew@webrtc.org · 10 years ago
  30. 7e2ad87 replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics. by andrew@webrtc.org · 10 years ago
  31. 63c5ce8 Add Opus support to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  32. 3894237 Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call. by stefan@webrtc.org · 10 years ago
  33. ed94bc8 Add stats for video: by asapersson@webrtc.org · 10 years ago
  34. 9b1042c Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already. by braveyao@webrtc.org · 10 years ago
  35. 5f53af3 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
  36. 6637388 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  37. 0eb4066 Update Android projects to API level 21. by kjellander@webrtc.org · 10 years ago
  38. 99f0de3 replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics. by andrew@webrtc.org · 10 years ago
  39. 46c5634 Add a WavReader counterpart to WavWriter. by andrew@webrtc.org · 10 years ago
  40. 796056b Update all .isolate files for the new format. by kjellander@webrtc.org · 10 years ago
  41. 4ed1b70 Update Android projects to API level 20. by kjellander@webrtc.org · 10 years ago
  42. cf59c6e Fix N7 camera aspect ratio. by glaznev@webrtc.org · 10 years ago
  43. f8586cc Build fix for MIPS32R6. by andrew@webrtc.org · 10 years ago
  44. 6aaaf9f Fix a name collision with Android libc++ by andrew@webrtc.org · 10 years ago
  45. ddb84aa Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  46. fbdea25 Configure A/V sync in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  47. 4ef173b Simplify bwe tests. by stefan@webrtc.org · 10 years ago
  48. a1f6cf7 Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
  49. 8a3acf6 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
  50. f37dc42 Add 15 fps support for Android devices with missing 15 fps camera mode. by glaznev@webrtc.org · 10 years ago
  51. d7062cc Creating a C++ wrapper class for VAD by henrik.lundin@webrtc.org · 10 years ago
  52. 78f89f1 Revert part of r7561, "Refactor audio conversion functions." by kwiberg@webrtc.org · 10 years ago
  53. bce1329 Refactor audio conversion functions. by andrew@webrtc.org · 10 years ago
  54. 7b5a896 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  55. 2ba45ee Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  56. 2d05389 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  57. 3c47df6 Remove unused code in overuse detector. by asapersson@webrtc.org · 10 years ago
  58. a999336 AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket by kwiberg@webrtc.org · 10 years ago
  59. ff8f833 Enable G.722 for Chromium builds by henrik.lundin@webrtc.org · 10 years ago
  60. dcfa54a Make an AudioEncoder subclass for Opus by kwiberg@webrtc.org · 10 years ago
  61. 06d5119 Make NSinst_t* const and rename to self in ns_core by aluebs@webrtc.org · 10 years ago
  62. 1a02faa move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  63. b136cd2 Make local functions static and dropWebRtcNs_ in ns_core by aluebs@webrtc.org · 10 years ago
  64. fc14046 Make all comments whole sentences in ns_core by aluebs@webrtc.org · 10 years ago
  65. 4f2aa08 scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. by henrike@webrtc.org · 10 years ago
  66. bc10410 audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" by bjornv@webrtc.org · 10 years ago
  67. 88b4afa common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 10 years ago
  68. e2cf507 Use neteq_unittest_tools in audio_decoder_unittests by henrik.lundin@webrtc.org · 10 years ago
  69. d18fd94 Fix double backslashes in incoming_video_stream.cc by perkj@webrtc.org · 10 years ago
  70. af3d97a Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
  71. 2e49acd Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago
  72. c91b433 Move (test) RtpFileReader to a lightweight target. by pbos@webrtc.org · 10 years ago
  73. b9d18d9 Move scoped_ptr "free" functions into the webrtc namespace. by andrew@webrtc.org · 10 years ago
  74. 2b687fb Upgrade our scoped_ptr copy to match Chromium's latest. by andrew@webrtc.org · 10 years ago
  75. c3f7292 Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile by henrik.lundin@webrtc.org · 10 years ago
  76. 185e7f9 isacfix: Refactor big-endian reading and writing by kwiberg@webrtc.org · 10 years ago
  77. a686801 Increase max trace message size to 1024 characters. by pbos@webrtc.org · 10 years ago
  78. 89ef054 Fix ::~LogMessage to print as a string. by pbos@webrtc.org · 10 years ago
  79. 3a8dbe3 Adding the subtool rtcBot report visualizer by houssainy@google.com · 10 years ago
  80. 2366875 Move min transmit bitrate to VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  81. f919668 Break out WebRtcNs_ComputeDdUpdate function in ns_core by aluebs@webrtc.org · 10 years ago
  82. 85900c9 Break out WebRtcNs_UpdateNoise function in ns_core by aluebs@webrtc.org · 10 years ago
  83. e5e6d52 Break out FFT function in ns_core by aluebs@webrtc.org · 10 years ago
  84. 101006b Break out ComputeSnr function in ns_core by aluebs@webrtc.org · 10 years ago
  85. e8c47e5 Adding three video conference bots test by houssainy@google.com · 10 years ago
  86. b7fdbc0 Adding file from test.webrtc.org domain to be downloaded by houssainy@google.com · 10 years ago
  87. b482152 Add macros and APIs for webrtc histograms. by asapersson@webrtc.org · 10 years ago
  88. dda5e80 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. by stefan@webrtc.org · 10 years ago
  89. 1546e3e Using the Unused turn configuration in two way test by houssainy@google.com · 10 years ago
  90. d7d29a3 Let video_loopback use internal VCM capturers. by pbos@webrtc.org · 10 years ago
  91. c080b41 NOTE: This code review based on the running issue: by houssainy@google.com · 10 years ago
  92. 45226bb Adding Two way video and audio streaming test to RtcBot by houssainy@google.com · 10 years ago
  93. cd3135a HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. by houssainy@google.com · 10 years ago
  94. 1c079a9 Make ReconfigureVideoEncoder use current bitrate. by pbos@webrtc.org · 10 years ago
  95. 0797c72 Disable TestVp8Impl.BaseUnitTest on MSan. by pbos@webrtc.org · 10 years ago
  96. 96b70f6 For FIR packet, payload length is zero, so SendToNetwork function is failing. by stefan@webrtc.org · 10 years ago
  97. 419897c Break out WebRtcNs_Windowing function in ns_core by aluebs@webrtc.org · 10 years ago
  98. faa322a Break out WebRtcNs_Energy function in ns_core by aluebs@webrtc.org · 10 years ago
  99. b3a6833 Break out WebRtcNs_IFFT function in ns_core by aluebs@webrtc.org · 10 years ago
  100. d696926 Break out WebRtcNs_UpdateBuffer function in ns_core by aluebs@webrtc.org · 10 years ago