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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / f1887a672d0b586d6c7834e1d3730f79cd719855
  1. f1887a6 TSan: Move suppressions to source file. by kjellander@webrtc.org · 10 years ago
  2. 1539b4c Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  3. 85e7fd1 Receiver bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 10 years ago
  4. f8b10b5 clock.h: Removed GUARDED_BY annotation as it breaks som builds. by henrike@webrtc.org · 10 years ago
  5. fb8cc91 Don't forward declare RWLockWrapper in clock.h by henrik.lundin@webrtc.org · 10 years ago
  6. eedc15c Fixes a bug causing NACKs to be dropped excessively at the send-side. by stefan@webrtc.org · 10 years ago
  7. 6851932 Bump version number to 3.55 by tnakamura@webrtc.org · 10 years ago
  8. 5f6bb3d fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. by henrike@webrtc.org · 10 years ago
  9. a8adf46 pkg-config-wrapper should not be run when build_nss is disabled (=0). by henrike@webrtc.org · 10 years ago
  10. fccff62 Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
  11. 99dfca3 This is to compare NetEq with various codecs under a shared packet loss pattern. by minyue@webrtc.org · 10 years ago
  12. 4f6ed09 Neon version of FilterFar() by bjornv@webrtc.org · 10 years ago
  13. ea61db0 Remove payload duplication in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  14. 4c611d1 Removing neteq decode lock and friends by henrik.lundin@webrtc.org · 10 years ago
  15. ac1ebeb Neon version of ScaleErrorSignal() by bjornv@webrtc.org · 10 years ago
  16. 10bb99f Annotating the rest of AcmGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  17. 865a242 Fix array declarations in aec_core.c by andrew@webrtc.org · 10 years ago
  18. 4e9d53d Annotating the rest of AudioCodingModuleImpl by henrik.lundin@webrtc.org · 10 years ago
  19. 97ce37c GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  20. a570079 Rebase webrtc/base with r6521 version of talk/base: by henrike@webrtc.org · 10 years ago
  21. 92e29b5 Disables tests that breaks Android bots by bjornv@webrtc.org · 10 years ago
  22. 31d4519 Roll chromium_revision 272489:277350 + fix sanitizer options by kjellander@webrtc.org · 10 years ago
  23. 289bbb3 GN: BUILD.gn for system_wrappers by kjellander@webrtc.org · 10 years ago
  24. 85729ac - Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper. by glaznev@webrtc.org · 10 years ago
  25. 72c05f7 Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread. by braveyao@webrtc.org · 10 years ago
  26. 620f146 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  27. 5295bdf Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. by jiayl@webrtc.org · 10 years ago
  28. 27b7548 Revert 6481 and 6482 by fgalligan@google.com · 10 years ago
  29. a4f29ab Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. by turaj@webrtc.org · 10 years ago
  30. 21e5713 Adding an empty constructor implementation to the AudioSink class by henrik.lundin@webrtc.org · 10 years ago
  31. 4095524 Changes to tests and tools in audio_processing. by bjornv@webrtc.org · 10 years ago
  32. f3db5dc Ensure that the start bitrate can be set multiple times. by stefan@webrtc.org · 10 years ago
  33. 445dc3c Adding test::AudioSink interface and derived classes by henrik.lundin@webrtc.org · 10 years ago
  34. 619b1a3 Fixes and re-enables tests disabled on Android by bjornv@webrtc.org · 10 years ago
  35. 8023ed2 Update webrtc to fix unpack_lib expansion. by fgalligan@google.com · 10 years ago
  36. 0cd9bb6 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  37. 7afb106 Neon version of FilterAdaptation() by bjornv@webrtc.org · 10 years ago
  38. f158d83 Update PacketSource and RtpFileSource by henrik.lundin@webrtc.org · 10 years ago
  39. 4bc91e2 Revert "Restore ptypes.txt file" by henrik.lundin@webrtc.org · 10 years ago
  40. edb557a Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 10 years ago
  41. b826558 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  42. 81bed12 Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 10 years ago
  43. b103f78 Add high perf mode to VP8 by niklas.enbom@webrtc.org · 10 years ago
  44. 91ccda5 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 10 years ago
  45. 57cbe42 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 10 years ago
  46. eec4302 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 10 years ago
  47. 2b152c1 Initial GN work for WebRTC by kjellander@webrtc.org · 10 years ago
  48. c551c3a Restore ptypes.txt file by henrik.lundin@webrtc.org · 10 years ago
  49. 7706520 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
  50. 9ed1dc4 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. by minyue@webrtc.org · 10 years ago
  51. f43487f Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. by henrike@webrtc.org · 10 years ago
  52. b929c5f Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  53. 5451d02 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  54. 341a2b3 Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  55. 2623b24 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 10 years ago
  56. e26a6fe Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 10 years ago
  57. 68457c7 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  58. 212705c Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  59. 226b500 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  60. caabc89 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
  61. d324c9c Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  62. 67230d1 Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
  63. 0f4e394 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  64. d03c245 Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
  65. 73d3f0d json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. by henrike@webrtc.org · 10 years ago
  66. ab25d49 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  67. e4757d1 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  68. 32f4c69 Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 10 years ago
  69. 50de2c1 Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 10 years ago
  70. a78c821 Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  71. 76e4554 Revert 6405 "Update generated asm offsets scripts." by henrike@webrtc.org · 10 years ago
  72. 48bbbc4 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  73. 5441ac0 Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  74. 3c20c98 Reland: Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  75. 508e748 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  76. c520f10 Revert 6395 "Making WebRTC able to play and record audio to file..." by minyue@webrtc.org · 10 years ago
  77. 15ce3f4 Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  78. f1deeba Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  79. 56a20ad common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  80. 344cbaf common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix by bjornv@webrtc.org · 10 years ago
  81. 84ba168 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 10 years ago
  82. 5f11104 Delete last file in neteq4 folder by henrik.lundin@webrtc.org · 10 years ago
  83. 76e47da MIPS optimizations for ISAC (patch #1) by andrew@webrtc.org · 10 years ago
  84. a76a757 Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 10 years ago
  85. 7d65bab Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  86. c3dd427 Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  87. d7d7cbc Create a joint encoder/decoder wrapper for iSAC in ACM by henrik.lundin@webrtc.org · 10 years ago
  88. e5e66d5 Add thread annotations to AcmReceiver by henrik.lundin@webrtc.org · 10 years ago
  89. da8b3f8 Make some methods in Clock class const declared by henrik.lundin@webrtc.org · 10 years ago
  90. 8ee3c9f Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
  91. e5035b3 Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
  92. 29fcd0f Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
  93. 75cb870 Multi-threaded unit test for Audio Coding Module using iSAC by henrik.lundin@webrtc.org · 10 years ago
  94. 80a43de audio_processing: Forces extended filter to be used in splitting filter test. by bjornv@webrtc.org · 10 years ago
  95. 823db07 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
  96. c08d93f Re-enable AudioCodingModuleMtTest again by henrik.lundin@webrtc.org · 10 years ago
  97. 0a84f06 Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." by fischman@webrtc.org · 10 years ago
  98. 691ee09 Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. by jiayl@webrtc.org · 10 years ago
  99. 131f703 AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. by fischman@webrtc.org · 10 years ago
  100. 95d222c Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. by fischman@webrtc.org · 10 years ago