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gerrit-public.fairphone.software / platform / external / webrtc / eb66e800d1f5f74ab366715d2618fbede8cf3e12
  1. f09e09c VoE: Remove unused interfaces by Jelena Marusic · 9 years ago
  2. 32c2023 Attempt at fixing error on the Chrome Windows FYI bots. by Tommi · 9 years ago
  3. 905495c Introduce NetEq::Config::ToString and use it in NetEq's constructor by Henrik Lundin · 9 years ago
  4. e982a70 PRESUBMIT: Fix typo. by Henrik Kjellander · 9 years ago
  5. 54be3e0 Remove some WebRtcVideoEngine2 unittest stubs. by Peter Boström · 9 years ago
  6. d8399e6 Also provide sample rate when registering decoders by Karl Wiberg · 9 years ago
  7. 323b132 Protect ACM decoder buffer in stereo. by Minyue · 9 years ago
  8. 57e5fd2 PRESUBMIT: Improve PyLint check and add GN format check. by Henrik Kjellander · 9 years ago
  9. 00aac5a Some cleanup for base/logging and base/stream.h by Tommi · 9 years ago
  10. 23edcff Move base/logging.* to rtc_base_approved. by Tommi · 9 years ago
  11. ee369e4 Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes by henrika · 9 years ago
  12. a26c4e5 Script to generate CL descriptions when rolling chromium_revision. by Henrik Kjellander · 9 years ago
  13. 0eefb4d Detach base/logging.* from base/stream.*. by Tommi · 9 years ago
  14. 469c2c0 Make Config::default_value leak instead of having an exit-time destructor. by Andrew MacDonald · 9 years ago
  15. 4bf12ea Revert "Fix sending wrong candidates down to transportchannel." by Alejandro Luebs · 9 years ago
  16. f65de84 Fix sending wrong candidates down to transportchannel. by Donald Curtis · 9 years ago
  17. 67b635a Fix simulcast_encoder_adapter giving full target_bitrate to the 2nd layer of any simulcast setup during InitEncode. by Noah Richards · 9 years ago
  18. e4cb4e9 Fix jitter buffer bug around out-of-order packets and non-RTX padding. by Noah Richards · 9 years ago
  19. 4774874 Enable AudioProcessing48kHzSupport by default by Alejandro Luebs · 9 years ago
  20. 3548dd2 Set local SSRCs on receivers added before senders. by Peter Boström · 9 years ago
  21. 367c868 AudioEncoderCng: Handle case where speech encoder is reset by Henrik Lundin · 9 years ago
  22. f761d10 Update NetEq Quality Test. by Minyue Li · 9 years ago
  23. 915df4f CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place. by Henrik Boström · 9 years ago
  24. 9a416bd Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2 by Fredrik Solenberg · 9 years ago
  25. 5af6d47 Code style change for quality_scaler. by jackychen · 9 years ago
  26. 98d8cf5 Hardware VP8 encoding: Use QP as metric for resize. by jackychen · 9 years ago
  27. 5fdcdf6 Enable ciphers to get ECDHE with NSS. by Joachim Bauch · 9 years ago
  28. 6f2ef74 Keep track of DTLS packet sizes to prevent partial reads. by Joachim Bauch · 9 years ago
  29. a3ba0c7 RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits by Magnus Jedvert · 9 years ago
  30. 36a1438 Remove ViEFrameProviderBase. by Peter Boström · 9 years ago
  31. af55ccc Add RtcpMuxPolicy support to PeerConnection. by Peter Thatcher · 9 years ago
  32. 02ff911 Feature merge request: Add support for iOS http proxy detection by Yuriy Shevchuk · 9 years ago
  33. 523183b Disables AudioDeviceTest.StartStopPlayout for Nexus 9 only by henrika · 9 years ago
  34. 280ed11 Roll gtest-parallel. by Peter Boström · 9 years ago
  35. 848d524 Revert "RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits" https://webrtc-codereview.appspot.com/47229004/ by Magnus Jedvert · 9 years ago
  36. 10022cd RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits by Magnus Jedvert · 9 years ago
  37. 78ae00e Remove default encoder/decoders. by Peter Boström · 9 years ago
  38. b302ad4 Remove unused VideoDecoder methods. by Peter Boström · 9 years ago
  39. 1a07a1e Solve data race in Pulse audio implementation. by Brave Yao · 9 years ago
  40. 8602a3d Cast to avoid char-interpretation of uint8_t in logs. by mflodman · 9 years ago
  41. 05c7605 Add resampling support in AudioBuffer::DeinterleaveFrom by Alejandro Luebs · 9 years ago
  42. 76b62ff Clean up now-unused code that was used for libpeerconnection.[so|dll]. by Tommi · 9 years ago
  43. fce3242 Remove linphonemediaengine.* by Fredrik Solenberg · 9 years ago
  44. 8eb76ff Make SHA1 computation thread-safe. by Sergey Ulanov · 9 years ago
  45. 5cdd702 Add tools/vim to .gitignore. by Andrew MacDonald · 9 years ago
  46. 9b2b402 Ensures that RECORD_AUDIO permission is required to start recording. by henrika · 9 years ago
  47. 5779d14 Avoids crash when StartRecording conflicts with existing recording application by henrika · 9 years ago
  48. c3f4dbc Remove rtp_rtcp/ dump functionality. by Peter Boström · 9 years ago
  49. ca667db Remove VCM debug recordings. by Peter Boström · 9 years ago
  50. 831c558 Allow setting maximum protocol version for SSL stream adapters. by Joachim Bauch · 9 years ago
  51. 664cdaf Replace assert() with static_assert() if the condition is evaluatable at by André Susano Pinto · 9 years ago
  52. 5ca688b Enable read-ahead on OpenSSL DTLS stream adapters. by Joachim Bauch · 9 years ago
  53. 931e658 Remove unnecessary dependencies for voe when building with include_internal_audio_device==0. by Tommi · 9 years ago
  54. cb7f8ce Clear ARM NEON flag by Andrew MacDonald · 9 years ago
  55. 4d71ede Add HW fallback option to software encoding. by Peter Boström · 9 years ago
  56. 97bce58 Disable the EXPECT_DEATH check in bitbuffer on Android by Donald E Curtis · 9 years ago
  57. bf560dd remove filelock which is now unused by Donald E Curtis · 9 years ago
  58. 17b889b Issue 4366: Adapted frames have wrong width and height and are cropped. by Guo-wei Shieh · 9 years ago
  59. 65de7d2 Add a link to tools/vim to use the Chromium YCM config with webrtc. by Andrew MacDonald · 9 years ago
  60. 5ece00f remove filelock which is now unused by Donald E Curtis · 9 years ago
  61. 2f5be9a Improve Android camera error handling. by Alex Glaznev · 9 years ago
  62. 68898a2 Remove AudioDeviceUtility. by Tommi · 9 years ago
  63. df0c05b Sort source file list for [rtc_]include_internal_audio_device. No code change. by Tommi · 9 years ago
  64. c2b63fe Adding Sony Xperia Z2 D6503 to HW AEC blacklist by henrika · 9 years ago
  65. 24e56e3 Fixes Chromium FYI build issue on Android. by henrika · 9 years ago
  66. ccb49e7 Remove Soundclip handling from libjingle. by Fredrik Solenberg · 9 years ago
  67. 1ab67ae Address the corner cases by Guo-wei Shieh · 9 years ago
  68. b92be45 Support 720P in portait as maximum on iOS. by Weiyong Yao · 9 years ago
  69. 8db8069 Change high frequency correction range by Alejandro Luebs · 9 years ago
  70. 3e95d3e Don't log warning for unexpected STUN binding responses. by Peter Thatcher · 9 years ago
  71. 79b2e06 Make the BlockDifference() functions return DiffInfo as their callers expect. by Peter Kasting · 9 years ago
  72. 2e7a098 Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. by Noah Richards · 9 years ago
  73. 7252a2b Add HW fallback option to software decoding. by Peter Boström · 9 years ago
  74. b261989 Adding support for OpenSL ES output in native WebRTC by henrika · 9 years ago
  75. 02c9b36 Roll gtest-parallel. by Peter Boström · 9 years ago
  76. 7e0c7d4 Add support for external encoders in ACM by Karl Wiberg · 9 years ago
  77. ea14f0a Move SetCurrentThreadName to platform_thread.* in rtc_base_approved, by Tommi · 9 years ago
  78. bd1bc47 Restructure decoder registration in ACM by Karl Wiberg · 9 years ago
  79. 9d8b71e Remove some dead code in ViEChannel. by Peter Boström · 9 years ago
  80. a6e883b Fix constant in SetCurrentThreadName. by André Susano Pinto · 9 years ago
  81. bebc690 Add platform_thread source files and move types from thread_checker_impl to there. by Tommi · 9 years ago
  82. 24ec128 Roll chromium_revision 5118a5b..1b9c098 (330060:330302) by Henrik Kjellander · 9 years ago
  83. a7d03ae Roll chromium_revision 62a5bb3..5118a5b (329063:330060) by Henrik Kjellander · 9 years ago
  84. 144d018 fix indent on tokenize_first function signatures by Donald Curtis · 9 years ago
  85. 42af6ca Add logging of "use candidate" and when we switch ICE "best" connections. by Peter Thatcher · 9 years ago
  86. b2d2623 Don't use rtc::LogCheckLevel, because it breaks Chrome. by Peter Thatcher · 9 years ago
  87. 1cf6f81 Add logging for sending and receiving STUN binding requests and TURN requests and responses. by Peter Thatcher · 9 years ago
  88. 37931c4 Stunprober interface, its implementation and a command line driver. by Guo-wei Shieh · 9 years ago
  89. 0e07f92 Split fmtp on semicolons not spaces as per RFC6871 by Donald Curtis · 9 years ago
  90. 20f3f94 Clear bitrate stats for unused SSRCs. by Peter Boström · 9 years ago
  91. 4cd6940 Enable -Wformat-security warning and cleanup GYP. by Henrik Kjellander · 9 years ago
  92. 39f2b0c Implemented video device info for iOS by Yuriy Shevchuk · 9 years ago
  93. a4463b2 Further updates to fix libjingle logging. by Tommi · 9 years ago
  94. 99eeee3 Fix logging in Chrome. by Tommi · 9 years ago
  95. 06c577f Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows. by Henrik Kjellander · 9 years ago
  96. 2013aec Propagating RTT from send-only channel to receive-only channel. by Minyue · 9 years ago
  97. 0703766 Fix issue where receive-side encoders are included in the padding bitrate. by Stefan Holmer · 9 years ago
  98. 9a63866 Move IncomingVideoFrames to common_video/. by Peter Boström · 9 years ago
  99. 4feb505 Remove VideoProcessing::ColorEnhancement. by Peter Boström · 9 years ago
  100. 5ec9985 Windows utility to setTheadName to help debugging. by André Susano Pinto · 9 years ago