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webrtc / src / webrtc / bed7b2fe7c6877de898f07ba165e08033d198f0e
  1. bed7b2f Put pseudotcp back because remoting uses it. by pthatcher@webrtc.org · 10 years ago
  2. 818b99e Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  3. 7027f05 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  4. bfca086 Moving encoded_bytes into EncodedInfo by henrik.lundin@webrtc.org · 10 years ago
  5. fb2415e Fix webrtc gn windows build. by kjellander@webrtc.org · 10 years ago
  6. 8ef3949 Removing manual test pages because they have been moved to github. by jansson@webrtc.org · 10 years ago
  7. f340352 Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
  8. ca9ea6b Move the downmixing out of AudioBuffer by aluebs@webrtc.org · 10 years ago
  9. cc2683c Adding DTX to WebRTC Opus wrapper (relanding). by minyue@webrtc.org · 10 years ago
  10. be573b5 Merge AEC changes. by pbos@webrtc.org · 10 years ago
  11. 834b475 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
  12. 28f5f3b Move isolate path into webrtc/build/android/test_runner.py by kjellander@webrtc.org · 10 years ago
  13. b7504b7 Make an AudioEncoder subclass for PCM16B by henrik.lundin@webrtc.org · 10 years ago
  14. 2a26de4 Make an AudioEncoder subclass for iSAC by kwiberg@webrtc.org · 10 years ago
  15. d13f42e Checking whether ACM uses codec internal or WebRTC DTX. by minyue@webrtc.org · 10 years ago
  16. 074735e DCHECK: Reference condition parameter in release builds by kwiberg@webrtc.org · 10 years ago
  17. dd50445 Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon by andrew@webrtc.org · 10 years ago
  18. 341822e Remove jitter_estimate_test.h by mflodman@webrtc.org · 10 years ago
  19. d2f6ea9 Support 48kHz in Noise Suppression by aluebs@webrtc.org · 10 years ago
  20. 764bd22 Remove CELT support from audio_coding. by pbos@webrtc.org · 10 years ago
  21. 6ef31ed Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. by asapersson@webrtc.org · 10 years ago
  22. dc4c7a2 Add AbsSendTime unittests to rampup_tests.cc. by pbos@webrtc.org · 10 years ago
  23. 62d5968 Cast payload type to int in logs. by asapersson@webrtc.org · 10 years ago
  24. 5aca070 Revert r7858 ("DCHECK: Reference condition parameter in release builds") by kwiberg@webrtc.org · 10 years ago
  25. b5ec993 DCHECK: Reference condition parameter in release builds by kwiberg@webrtc.org · 10 years ago
  26. ab05c0f Make an AudioEncoder subclass for comfort noise by henrik.lundin@webrtc.org · 10 years ago
  27. ada6e4e Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon. by andrew@webrtc.org · 10 years ago
  28. 54a94d7 Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon. by andrew@webrtc.org · 10 years ago
  29. 6c79878 Attempt to fix FYI bots. by tommi@webrtc.org · 10 years ago
  30. db5c754 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer by henrika@webrtc.org · 10 years ago
  31. 3d69237 Revert 7846 "Adding DTX to WebRTC Opus wrapper" by minyue@webrtc.org · 10 years ago
  32. 59cfd6d Add histograms for receive statistics: by asapersson@webrtc.org · 10 years ago
  33. 09299c2 Adding DTX to WebRTC Opus wrapper by minyue@webrtc.org · 10 years ago
  34. 32494ca Adding an codec interal CNG test in NetEq. by minyue@webrtc.org · 10 years ago
  35. 866b22b Merge VP8 changes. by pbos@webrtc.org · 10 years ago
  36. f41b165 Move the AudioDecoder interface out of NetEq by kwiberg@webrtc.org · 10 years ago
  37. 96568c2 Add video send bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  38. 55326d8 Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper. by kjellander@webrtc.org · 10 years ago
  39. 3e907c2 Remove no longer used video codec test framework. by stefan@webrtc.org · 10 years ago
  40. 44271fd Add AudioEncoder::Max10MsFramesInAPacket by henrik.lundin@webrtc.org · 10 years ago
  41. 72cfef7 Bugfix in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  42. a5f17c8 Change all system clock types to int64_t in bitrate_controller. by stefan@webrtc.org · 10 years ago
  43. dc0466a Add const qualifier to WebRtcPcm16b_Encode by henrik.lundin@webrtc.org · 10 years ago
  44. c10f0e6 ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable. by kwiberg@webrtc.org · 10 years ago
  45. db1e503 Make an AudioEncoder subclass for iLBC by kwiberg@webrtc.org · 10 years ago
  46. 0668dba Cleaned up real_fft APIs due to non-existing NEON code by bjornv@webrtc.org · 10 years ago
  47. 0261d88 Change type of nack_last_time_sent_full_ from uint32_t to int64_t. by asapersson@webrtc.org · 10 years ago
  48. b357f65 Adjust some parameters for VP9 tests. by marpan@webrtc.org · 10 years ago
  49. 871e148 Add codereview.settings to the /webrtc subdirectory by kjellander@webrtc.org · 10 years ago
  50. 04b5c53 Add support for parsing header only RTP dumps with bwe_rtp_play. by stefan@webrtc.org · 10 years ago
  51. 57a3a82 Merge remote bitrate estimator changes. by pbos@webrtc.org · 10 years ago
  52. 1c3ff4e Relanding r7807. by minyue@webrtc.org · 10 years ago
  53. 8a7700f Revert 7807 "Removing unused opus wrapper APIs." by minyue@webrtc.org · 10 years ago
  54. c90f9c2 Removing unused opus wrapper APIs. by minyue@webrtc.org · 10 years ago
  55. 5b66cc8 Redo the change of https://webrtc-codereview.appspot.com/30949004/ by guoweis@webrtc.org · 10 years ago
  56. cb86fa9 Revert "Implement GetState() for channel's connectivity check state." by guoweis@webrtc.org · 10 years ago
  57. 75b5d25 Implement GetState() for channel's connectivity check state. by guoweis@webrtc.org · 10 years ago
  58. b7ebe5b Adding WebRtcSpl_MaxAbsValueW16 intrinsics version by andrew@webrtc.org · 10 years ago
  59. 9e2103c add WebRtcIsacfix_AutocorrNeon's intrinsics version by andrew@webrtc.org · 10 years ago
  60. ec40887 Rename internal AudioEncoder::Encode method to EncodeInternal by henrik.lundin@webrtc.org · 10 years ago
  61. 26e77e1 Remove need for assembly offset generation in aecm and ns module. by andrew@webrtc.org · 10 years ago
  62. 4a32048 Revert r7798 ("Move the AudioDecoder interface out of NetEq") by kwiberg@webrtc.org · 10 years ago
  63. 5ded9f9 Move the AudioDecoder interface out of NetEq by kwiberg@webrtc.org · 10 years ago
  64. 3d1af07 Adding a duration printout to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  65. 0b55fd3 Add Android test runner script for WebRTC. by kjellander@webrtc.org · 10 years ago
  66. bb8b302 TurnPort should ignore STUN binding reponses when using shared socket. by jiayl@webrtc.org · 10 years ago
  67. 0ac3fd9 Adjust parameter in videoprocessor_integration_test for vp9. by marpan@webrtc.org · 10 years ago
  68. 3e39691 Simplify audio_buffer APIs by aluebs@webrtc.org · 10 years ago
  69. c428945 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. by marpan@webrtc.org · 10 years ago
  70. ba372c0 Remove -flax-vector-conversions flag for ARM NEON building. by andrew@webrtc.org · 10 years ago
  71. 422bafd Clear 2 unused functions in audio processing aecm module. by andrew@webrtc.org · 10 years ago
  72. f9aaa39 Adding a payload type to AudioEncoder objects by henrik.lundin@webrtc.org · 10 years ago
  73. f088f16 AudioEncoder subclass for G722 by kwiberg@webrtc.org · 10 years ago
  74. 512f947 Roll chromium_revision 309cf65..24b4c73 by kjellander@webrtc.org · 10 years ago
  75. 44b7577 Use c++11 features in webrtc/base/network.cc as a test to see if we can use them. by pthatcher@webrtc.org · 10 years ago
  76. 29ea7da Set simulcastIdx field to zero even if it has no meaning. by andresp@webrtc.org · 10 years ago
  77. ee9497c Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  78. 5576859 Adding EncodedInfo struct to AudioEncoder::Encode by henrik.lundin@webrtc.org · 10 years ago
  79. 35c0a57 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat by henrik.lundin@webrtc.org · 10 years ago
  80. 81a1158 Add test NetEqDecodingTest.CngFirst by henrik.lundin@webrtc.org · 10 years ago
  81. 3b73273 Adding a new test helper RtpFileWriter and use it in RTPcat by henrik.lundin@webrtc.org · 10 years ago
  82. 173e417 Add framerate for complete received frames to histogram stats: by asapersson@webrtc.org · 10 years ago
  83. 03bee64 Make bands vector in SplittingFilter Analysis const by aluebs@webrtc.org · 10 years ago
  84. 3550277 Move ChannelBuffer class to channel_buffer file by aluebs@webrtc.org · 10 years ago
  85. f800961 Remove unused RtpStatistics struct. by pbos@webrtc.org · 10 years ago
  86. 2755e82 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  87. 4d9d595 Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands by aluebs@webrtc.org · 10 years ago
  88. 4ab1d2b Fix an ASSERT that fires in a browser test for renegotiation. by jiayl@webrtc.org · 10 years ago
  89. 1b53688 Enabling building with NEON on ARM64 by andrew@webrtc.org · 10 years ago
  90. 101bf4d Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
  91. 5e3f6b4 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  92. e36c5bc Add back EXPECT_TRUEs. by pbos@webrtc.org · 10 years ago
  93. 39f38e3 Reenable GetStats test. by pbos@webrtc.org · 10 years ago
  94. b1bd389 Add wav output capability to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
  95. db77c84 Add new test for VP8 packetizer to test tight partitions by henrik.lundin@webrtc.org · 10 years ago
  96. 45c5e1f OWNERS: Remove tomasl@ and mallinath@ by kjellander@webrtc.org · 10 years ago
  97. 5bcd6b4 Simplifying VideoReceiver and JitterBuffer. by pbos@webrtc.org · 10 years ago
  98. 0532dd7 Use vector of CSRCs for DeliverFrame & SetCSRCs. by pbos@webrtc.org · 10 years ago
  99. a7fe477 Build fix for MIPS Android Webview build. by andrew@webrtc.org · 10 years ago
  100. 41266d1 Update mock_frame_dropper.h to use size_t by kjellander@webrtc.org · 10 years ago