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webrtc / src.git / 02c9b3673327d7e03f8e3ea787105c63a322c29e / webrtc
  1. 7e0c7d4 Add support for external encoders in ACM by Karl Wiberg · 10 years ago
  2. ea14f0a Move SetCurrentThreadName to platform_thread.* in rtc_base_approved, by Tommi · 10 years ago
  3. bd1bc47 Restructure decoder registration in ACM by Karl Wiberg · 10 years ago
  4. 9d8b71e Remove some dead code in ViEChannel. by Peter Boström · 10 years ago
  5. a6e883b Fix constant in SetCurrentThreadName. by André Susano Pinto · 10 years ago
  6. bebc690 Add platform_thread source files and move types from thread_checker_impl to there. by Tommi · 10 years ago
  7. 144d018 fix indent on tokenize_first function signatures by Donald Curtis · 10 years ago
  8. 42af6ca Add logging of "use candidate" and when we switch ICE "best" connections. by Peter Thatcher · 10 years ago
  9. b2d2623 Don't use rtc::LogCheckLevel, because it breaks Chrome. by Peter Thatcher · 10 years ago
  10. 1cf6f81 Add logging for sending and receiving STUN binding requests and TURN requests and responses. by Peter Thatcher · 10 years ago
  11. 37931c4 Stunprober interface, its implementation and a command line driver. by Guo-wei Shieh · 10 years ago
  12. 0e07f92 Split fmtp on semicolons not spaces as per RFC6871 by Donald Curtis · 10 years ago
  13. 20f3f94 Clear bitrate stats for unused SSRCs. by Peter Boström · 10 years ago
  14. 4cd6940 Enable -Wformat-security warning and cleanup GYP. by Henrik Kjellander · 10 years ago
  15. 39f2b0c Implemented video device info for iOS by Yuriy Shevchuk · 10 years ago
  16. a4463b2 Further updates to fix libjingle logging. by Tommi · 10 years ago
  17. 99eeee3 Fix logging in Chrome. by Tommi · 10 years ago
  18. 06c577f Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows. by Henrik Kjellander · 10 years ago
  19. 2013aec Propagating RTT from send-only channel to receive-only channel. by Minyue · 10 years ago
  20. 0703766 Fix issue where receive-side encoders are included in the padding bitrate. by Stefan Holmer · 10 years ago
  21. 9a63866 Move IncomingVideoFrames to common_video/. by Peter Boström · 10 years ago
  22. 4feb505 Remove VideoProcessing::ColorEnhancement. by Peter Boström · 10 years ago
  23. 5ec9985 Windows utility to setTheadName to help debugging. by André Susano Pinto · 10 years ago
  24. 9b9f1c4 Remove basictypes.h dependency from bitbuffer. by Noah Richards · 10 years ago
  25. e235714 Guard new protobuf target with enable_protobuf==1. by Andrew MacDonald · 10 years ago
  26. 300eeb6 Remove VideoEngine interfaces. by Peter Boström · 10 years ago
  27. 8171735 Add NetEqIlbcQualityTest by Henrik Lundin · 10 years ago
  28. df66453 Remove FPS->kilo-FPS conversion in VideoSender. by Peter Boström · 10 years ago
  29. e5ff00a Add NetEqPcmuQualityTest by Henrik Lundin · 10 years ago
  30. fade179 Remove leaking aecdump testfiles. by Peter Boström · 10 years ago
  31. 075bb8d Fix race in AudioCodingModuleImpl::Add10MsData() by Karl Wiberg · 10 years ago
  32. 1b794d5 Switch to use SHA-256 for certificates / fingerprints. by Joachim Bauch · 10 years ago
  33. cb3e8fe Increase the tolerance in NetEq's DelayManagerTest a notch by Henrik Lundin · 10 years ago
  34. 64dad83 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  35. 092041c Setting OPUS_SIGNAL_VOICE when enable DTX. by Minyue Li · 10 years ago
  36. 9f7908e Roll chromium_revision ec5b768..62a5bb3 (328242:329063) by Henrik Kjellander · 10 years ago
  37. 242e22b Refactor RTCP sender by Erik Språng · 10 years ago
  38. 1f62923 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  39. fd32f35 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  40. 54adb28 mac: Explicitly redeclare methods only available on 10.7+. by Jiayang Liu · 10 years ago
  41. cdb47a4 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  42. 45553ae Remove VideoEngine interface usage from new API. by Peter Boström · 10 years ago
  43. 208a229 Adding a new constraint to set NetEq buffer capacity from peerconnection by Henrik Lundin · 10 years ago
  44. 83b5c05 Modify NetEqQualityTest by Henrik Lundin · 10 years ago
  45. cb05b72 Add WAV and arbitrary geometry support to nlbf test. by Andrew MacDonald · 10 years ago
  46. 2ea71c3 Replace ACMGenericCodec with CodecOwner and AudioEncoderMutable by Karl Wiberg · 10 years ago
  47. 53d0dc3 Wire up RTT to send-side GCC and TCP. by Stefan Holmer · 10 years ago
  48. dcccab3 New interface: AudioEncoderMutable by Karl Wiberg · 10 years ago
  49. c81591d NADA's proposal from Cisco. by Cesar Magalhaes · 10 years ago
  50. f353dd5 VoE: cleanup VoENetwork implementation by Jelena Marusic · 10 years ago
  51. 1ff218f audio_processing/aec: Do not scale target delay at startup when on Android by Bjorn Volcker · 10 years ago
  52. 532531b audio_processing/delay_estimator: Always update robust validation statistics by Bjorn Volcker · 10 years ago
  53. 40a6d59 audio_processing/tests: Adds a flag to unpack input data to text file by Bjorn Volcker · 10 years ago
  54. 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
  55. f242e66 Replace asm NEON function by intrinsics implementation on ARMv7 by Zhongwei Yao · 10 years ago
  56. 589699e Fix bug in transform_neon.c in iSAC codec. by Zhongwei Yao · 10 years ago
  57. 5cb9ce4 Remove ViECodec usage in VideoSendStream. by Peter Boström · 10 years ago
  58. ab00404 VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument by Magnus Jedvert · 10 years ago
  59. 01b4888 Use padding to achieve bitrate probing if the initial key frame has too few packets. by Stefan Holmer · 10 years ago
  60. c56ac1e rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  61. f75f0cf Enable GoogleWifiTrace3Mbps simulations. by Stefan Holmer · 10 years ago
  62. 0d26605 VoE: apply new style guide on VoE interfaces and their implementations by Jelena Marusic · 10 years ago
  63. 79c1433 Delete VoiceChannelTransport before deleting Channel in voe_cmd_test by Minyue Li · 10 years ago
  64. 0b15445 VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/ by Jelena Marusic · 10 years ago
  65. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 10 years ago
  66. 4eddf18 Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle. by Peter Thatcher · 10 years ago
  67. 8a6680e Remove base/move.h (no one uses it anymore) by Karl Wiberg · 10 years ago
  68. cbf0927 Revert "rtc::Buffer: Remove backwards compatibility band-aids" by Karl Wiberg · 10 years ago
  69. 9e1a6d7 rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  70. ff019b0 Move rtc::AtomicOps to webrtc/base/atomicops.h. by Peter Boström · 10 years ago
  71. f16fcbe Remove ViECapture usage in VideoSendStream. by Peter Boström · 10 years ago
  72. 46bd31b VoE: VoENetwork unit test by Jelena Marusic · 10 years ago
  73. 3cfa756 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz by Bjorn Volcker · 10 years ago
  74. adf89b7e Added SetBitRate function to VoE API to allow changing the audio bitrate. by Ivo Creusen · 10 years ago
  75. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  76. dea11f9 Add per flow throughput and delay metrics. by Stefan Holmer · 10 years ago
  77. 94cc1fe Remove ViEImageProcess usage in VideoSendStream. by Peter Boström · 10 years ago
  78. 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
  79. 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
  80. 80154f6 Set correct .type directive for asm functions. by Wei Zhong · 10 years ago
  81. 019087f Add safeguards against signalling peer-reflexive candidates. by Peter Thatcher · 10 years ago
  82. 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
  83. 88de479 AudioEncoderIsac: Print error code if CHECK for successful encoding fails by Karl Wiberg · 10 years ago
  84. bcbcd84 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset. by Stefan Holmer · 10 years ago
  85. 9d657cf Fix dangling pointer in screenshare_loopback by Erik Språng · 10 years ago
  86. beb9798 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool by Bjorn Volcker · 10 years ago
  87. ddbddbd Remove ViENetwork usage in VideoSendStream. by Peter Boström · 10 years ago
  88. 038df3c Remove ViEExternalCodec usage in VideoSendStream. by Peter Boström · 10 years ago
  89. 4a9cb6b Prevent zero-timestamps in captured_frame_. by Peter Boström · 10 years ago
  90. 143cec1 Set correct encoder-specific settings for vpx in the new API. by Erik Språng · 10 years ago
  91. e8a197b Enable isac NEON building on Aarch64 by Zhongwei Yao · 10 years ago
  92. d7e5c44 STUN allocation should not be disabled when using shared port and TURN servers are provided. by Jiayang Liu · 10 years ago
  93. 5a92aa8 Add 3-band filter-bank implementation by Alejandro Luebs · 10 years ago
  94. 494f209 Move CriticalSection into rtc_base_approved. by Tommi · 10 years ago
  95. 59d91dc Remove ViERTP_RTCP usage in VideoSendStream. by Peter Boström · 10 years ago
  96. e6cefb6 GYP variables for building expat, icu, libsrtp, usrsctp by Henrik Kjellander · 10 years ago
  97. 61be2a4 Clean up RTCPSender. by Erik Språng · 10 years ago
  98. 3c391cb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. by Åsa Persson · 10 years ago
  99. 52ef9d7 Stop IncomingVideoStream on delete. by Peter Boström · 10 years ago
  100. 23dc68e Add the rtc_build_openmax_dl variable to the GN build. by Andrew MacDonald · 10 years ago