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gerrit-public.fairphone.software / platform / external / webrtc / 0e07f92043b333acfdaed8f22da5df903a70e0e9
  1. 0e07f92 Split fmtp on semicolons not spaces as per RFC6871 by Donald Curtis · 10 years ago
  2. 20f3f94 Clear bitrate stats for unused SSRCs. by Peter Boström · 10 years ago
  3. 4cd6940 Enable -Wformat-security warning and cleanup GYP. by Henrik Kjellander · 10 years ago
  4. 39f2b0c Implemented video device info for iOS by Yuriy Shevchuk · 10 years ago
  5. a4463b2 Further updates to fix libjingle logging. by Tommi · 10 years ago
  6. 99eeee3 Fix logging in Chrome. by Tommi · 10 years ago
  7. 06c577f Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows. by Henrik Kjellander · 10 years ago
  8. 2013aec Propagating RTT from send-only channel to receive-only channel. by Minyue · 10 years ago
  9. 0703766 Fix issue where receive-side encoders are included in the padding bitrate. by Stefan Holmer · 10 years ago
  10. 9a63866 Move IncomingVideoFrames to common_video/. by Peter Boström · 10 years ago
  11. 4feb505 Remove VideoProcessing::ColorEnhancement. by Peter Boström · 10 years ago
  12. 5ec9985 Windows utility to setTheadName to help debugging. by André Susano Pinto · 10 years ago
  13. 9b9f1c4 Remove basictypes.h dependency from bitbuffer. by Noah Richards · 10 years ago
  14. e235714 Guard new protobuf target with enable_protobuf==1. by Andrew MacDonald · 10 years ago
  15. 300eeb6 Remove VideoEngine interfaces. by Peter Boström · 10 years ago
  16. 8171735 Add NetEqIlbcQualityTest by Henrik Lundin · 10 years ago
  17. df66453 Remove FPS->kilo-FPS conversion in VideoSender. by Peter Boström · 10 years ago
  18. e5ff00a Add NetEqPcmuQualityTest by Henrik Lundin · 10 years ago
  19. fade179 Remove leaking aecdump testfiles. by Peter Boström · 10 years ago
  20. 075bb8d Fix race in AudioCodingModuleImpl::Add10MsData() by Karl Wiberg · 10 years ago
  21. 1b794d5 Switch to use SHA-256 for certificates / fingerprints. by Joachim Bauch · 10 years ago
  22. cb3e8fe Increase the tolerance in NetEq's DelayManagerTest a notch by Henrik Lundin · 10 years ago
  23. 67c9df7 Base NACK on send codecs. by Peter Boström · 10 years ago
  24. 126c03e Base decision to send REMB on send codecs. by Peter Boström · 10 years ago
  25. 64dad83 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  26. 092041c Setting OPUS_SIGNAL_VOICE when enable DTX. by Minyue Li · 10 years ago
  27. 9f7908e Roll chromium_revision ec5b768..62a5bb3 (328242:329063) by Henrik Kjellander · 10 years ago
  28. 242e22b Refactor RTCP sender by Erik Språng · 10 years ago
  29. 1f62923 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  30. fd32f35 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  31. 54adb28 mac: Explicitly redeclare methods only available on 10.7+. by Jiayang Liu · 10 years ago
  32. 4c277bb Add basic SCTP packet logging. by Lally Singh · 10 years ago
  33. cdb47a4 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  34. 45553ae Remove VideoEngine interface usage from new API. by Peter Boström · 10 years ago
  35. 208a229 Adding a new constraint to set NetEq buffer capacity from peerconnection by Henrik Lundin · 10 years ago
  36. 83b5c05 Modify NetEqQualityTest by Henrik Lundin · 10 years ago
  37. cb05b72 Add WAV and arbitrary geometry support to nlbf test. by Andrew MacDonald · 10 years ago
  38. d3ddc1b Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/. by Fredrik Solenberg · 10 years ago
  39. e444a3d WebRtcVoiceEngine: Get rid of unnecessary template base class. by Fredrik Solenberg · 10 years ago
  40. aaf8ff2 WebRtcVoiceEngine: virtual to override + git cl format. by Fredrik Solenberg · 10 years ago
  41. 6179b89 Remove unused API on WebRtcVoiceEngine. by Fredrik Solenberg · 10 years ago
  42. 2ea71c3 Replace ACMGenericCodec with CodecOwner and AudioEncoderMutable by Karl Wiberg · 10 years ago
  43. 53d0dc3 Wire up RTT to send-side GCC and TCP. by Stefan Holmer · 10 years ago
  44. 4b60c73 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE. by Fredrik Solenberg · 10 years ago
  45. dcccab3 New interface: AudioEncoderMutable by Karl Wiberg · 10 years ago
  46. 81ea54e Remove WebRtcVideoEngine. by Peter Boström · 10 years ago
  47. ccfc939 Reinterpret AudioOption delay_agnostic_aec to override HW-AEC by Bjorn Volcker · 10 years ago
  48. c81591d NADA's proposal from Cisco. by Cesar Magalhaes · 10 years ago
  49. f353dd5 VoE: cleanup VoENetwork implementation by Jelena Marusic · 10 years ago
  50. 1ff218f audio_processing/aec: Do not scale target delay at startup when on Android by Bjorn Volcker · 10 years ago
  51. 532531b audio_processing/delay_estimator: Always update robust validation statistics by Bjorn Volcker · 10 years ago
  52. 40a6d59 audio_processing/tests: Adds a flag to unpack input data to text file by Bjorn Volcker · 10 years ago
  53. 9695d85 Added VP9FrameBufferPool, a memory pool that is shared between libvpx and webrtc. Using the VP9 codec, the libvpx decoder will obtain its buffers from our memory pool. This lets us reuse the same buffers for our I420VideoFrames and not have to copy a frame for every decode (from libvpx buffers to webrtc/I420VideoFrame buffers). by Henrik Boström · 10 years ago
  54. f242e66 Replace asm NEON function by intrinsics implementation on ARMv7 by Zhongwei Yao · 10 years ago
  55. 507a550 Delete auto-roll script since moved into Chromium. by Henrik Kjellander · 10 years ago
  56. 589699e Fix bug in transform_neon.c in iSAC codec. by Zhongwei Yao · 10 years ago
  57. 57cc74e iOS camera switching video capturer. by Zeke Chin · 10 years ago
  58. 5cb9ce4 Remove ViECodec usage in VideoSendStream. by Peter Boström · 10 years ago
  59. ab00404 VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument by Magnus Jedvert · 10 years ago
  60. 01b4888 Use padding to achieve bitrate probing if the initial key frame has too few packets. by Stefan Holmer · 10 years ago
  61. 78c8bbf Roll chromium_revision 0cb2549..ec5b768 (327252:328242) by Henrik Kjellander · 10 years ago
  62. c56ac1e rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  63. f75f0cf Enable GoogleWifiTrace3Mbps simulations. by Stefan Holmer · 10 years ago
  64. 0d26605 VoE: apply new style guide on VoE interfaces and their implementations by Jelena Marusic · 10 years ago
  65. 79c1433 Delete VoiceChannelTransport before deleting Channel in voe_cmd_test by Minyue Li · 10 years ago
  66. 0b15445 VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/ by Jelena Marusic · 10 years ago
  67. e433c0e Restore back verbosity logging for camera captured frame. by Alex Glaznev · 10 years ago
  68. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
  69. cac1b38 Expose RTCConfiguration to java JNI and add an option to disable TCP by Jiayang Liu · 10 years ago
  70. 4eddf18 Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle. by Peter Thatcher · 10 years ago
  71. 8a6680e Remove base/move.h (no one uses it anymore) by Karl Wiberg · 10 years ago
  72. cbf0927 Revert "rtc::Buffer: Remove backwards compatibility band-aids" by Karl Wiberg · 10 years ago
  73. 9e1a6d7 rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  74. ff019b0 Move rtc::AtomicOps to webrtc/base/atomicops.h. by Peter Boström · 10 years ago
  75. f16fcbe Remove ViECapture usage in VideoSendStream. by Peter Boström · 10 years ago
  76. 46bd31b VoE: VoENetwork unit test by Jelena Marusic · 10 years ago
  77. 3cfa756 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz by Bjorn Volcker · 10 years ago
  78. efbde37 Don't use CPU adaptation for screen content in the new API. by Erik Språng · 10 years ago
  79. adf89b7 Added SetBitRate function to VoE API to allow changing the audio bitrate. by Ivo Creusen · 10 years ago
  80. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  81. 10ba3ee Roll chromium_revision a12e1e1..0cb2549 (326495:327252) by Henrik Kjellander · 10 years ago
  82. dea11f9 Add per flow throughput and delay metrics. by Stefan Holmer · 10 years ago
  83. 94cc1fe Remove ViEImageProcess usage in VideoSendStream. by Peter Boström · 10 years ago
  84. c444de6 Make setup_links.py handle non-link directories during cleanup by Henrik Kjellander · 10 years ago
  85. 1ba344a Adds a MediaConstraint for the AudioOption aec_dump by Bjorn Volcker · 10 years ago
  86. 97f13c5 Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0. by Noah Richards · 10 years ago
  87. 86153c2 Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data. by Noah Richards · 10 years ago
  88. 80154f6 Set correct .type directive for asm functions. by Wei Zhong · 10 years ago
  89. faa6d07 Remove a few verbose log messages from webrtcvideoengine2. by Alex Glaznev · 10 years ago
  90. 019087f Add safeguards against signalling peer-reflexive candidates. by Peter Thatcher · 10 years ago
  91. ae33134 Always specify current OS when syncing Chromium. by Henrik Kjellander · 10 years ago
  92. 8786f63 Roll gtest-parallel. by Peter Boström · 10 years ago
  93. 31dc737 Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel. by Stefan Holmer · 10 years ago
  94. 88de479 AudioEncoderIsac: Print error code if CHECK for successful encoding fails by Karl Wiberg · 10 years ago
  95. bcbcd84 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset. by Stefan Holmer · 10 years ago
  96. 9d657cf Fix dangling pointer in screenshare_loopback by Erik Språng · 10 years ago
  97. beb9798 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool by Bjorn Volcker · 10 years ago
  98. ddbddbd Remove ViENetwork usage in VideoSendStream. by Peter Boström · 10 years ago
  99. 038df3c Remove ViEExternalCodec usage in VideoSendStream. by Peter Boström · 10 years ago
  100. 4a9cb6b Prevent zero-timestamps in captured_frame_. by Peter Boström · 10 years ago