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android-review.linaro.org / platform / external / chromium_org / third_party / webrtc / refs/heads/l-preview
  1. aecf494 merge in lmp-preview-dev history after reset to 2dc33605704d570ebd2ab5f0428ea94672ce6a7e by The Android Automerger · 10 years ago l-preview android-l-preview_r2
  2. e560614 merge in lmp-preview-dev history after reset to 2dc33605704d570ebd2ab5f0428ea94672ce6a7e by The Android Automerger · 10 years ago
  3. 2dc3360 Update makefiles after merge by Torne (Richard Coles) · 10 years ago lollipop-cts-release lollipop-dev lollipop-release lollipop-wear-release android-5.0.0_r1 android-5.0.0_r2 android-5.0.0_r3 android-5.0.0_r4 android-5.0.0_r5 android-5.0.0_r5.1 android-5.0.0_r6 android-5.0.0_r7 android-5.0.1_r1 android-5.0.2_r1 android-5.0.2_r3 android-cts-5.0_r3 android-cts-5.0_r4 android-cts-5.0_r5 android-cts-5.0_r6 android-cts-5.0_r7 android-cts-5.0_r8 android-cts-5.0_r9 android-wear-5.0.0_r1
  4. fc979d2 Merge from Chromium at DEPS revision 37.0.2062.68 by Bo Liu · 10 years ago
  5. df09699 Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 14b8f01ca3c95e3f10a141f63c3250b38cf5433c by Bo Liu · 10 years ago
  6. 1ab7dd3 Merge from Chromium at DEPS revision 37.0.2062.52 by Bo Liu · 10 years ago
  7. a7f46bc Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 939b8c943d0198c81702beec3d7faa152b8a29a7 by Bo Liu · 10 years ago
  8. 14b8f01 Merge r6774 to branch 3.55. by sprang@webrtc.org · 10 years ago
  9. 939b8c9 Merge r6571 and r6572 to the 3.55 branch. by stefan@webrtc.org · 10 years ago
  10. 1515b4c Merge from Chromium at DEPS revision 37.0.2062.21 by Bo Liu · 10 years ago
  11. f58511d Update makefiles after merge of Chromium at 37.0.2062.21 by Bo Liu · 10 years ago
  12. d876846 Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 1b4ed025b291d5d226ee45f1f1211becd39a3560 by Bo Liu · 10 years ago
  13. 1b4ed02 Merge r6544 to 3.55 branch. by tnakamura@webrtc.org · 10 years ago
  14. 143e70b Merge from Chromium at DEPS revision 37.0.2062.10 by Torne (Richard Coles) · 10 years ago
  15. 8e557e7 Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at b5f5e9089dca995c3d4a6fe0c266d19b8a088b92 by Torne (Richard Coles) · 10 years ago
  16. b5f5e90 Create WebRTC 3.55 branch from trunk@6496 by tnakamura@webrtc.org · 10 years ago
  17. 65b8a71 Merge from Chromium at DEPS revision 278856 by Torne (Richard Coles) · 10 years ago
  18. b0cf5e1 Update makefiles after merge of Chromium at 278856 by Torne (Richard Coles) · 10 years ago
  19. c5e87ee Merge from Chromium at DEPS revision 278205 by Torne (Richard Coles) · 10 years ago
  20. c497bcd Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 68f4c7b51ec6434b302de9e97ee01f5ccdb48aa2 by Android Chromium Automerger · 10 years ago
  21. 68f4c7b Revert 6481 and 6482 by fgalligan@google.com · 10 years ago
  22. c7a2c99 Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. by turaj@webrtc.org · 10 years ago
  23. 035f764 Adding an empty constructor implementation to the AudioSink class by henrik.lundin@webrtc.org · 10 years ago
  24. 8cda294 Changes to tests and tools in audio_processing. by bjornv@webrtc.org · 10 years ago
  25. b9bd8c8 Ensure that the start bitrate can be set multiple times. by stefan@webrtc.org · 10 years ago
  26. 5c69a9f Adding test::AudioSink interface and derived classes by henrik.lundin@webrtc.org · 10 years ago
  27. 341671f Fixes and re-enables tests disabled on Android by bjornv@webrtc.org · 10 years ago
  28. ad3bcf4 Update makefiles after merge of Chromium at 278252 by Android Chromium Automerger · 10 years ago
  29. 852ce03 Update webrtc to fix unpack_lib expansion. by fgalligan@google.com · 10 years ago
  30. dfaba91 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  31. 685eb96 Neon version of FilterAdaptation() by bjornv@webrtc.org · 10 years ago
  32. 82f4b96 Update PacketSource and RtpFileSource by henrik.lundin@webrtc.org · 10 years ago
  33. 743f486 Revert "Restore ptypes.txt file" by henrik.lundin@webrtc.org · 10 years ago
  34. 79ceb8c Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 10 years ago
  35. 6f1646c Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  36. 7e87822 Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 10 years ago
  37. 50d455e Add high perf mode to VP8 by niklas.enbom@webrtc.org · 10 years ago
  38. a487491 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 10 years ago
  39. fd6b7a5 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 10 years ago
  40. 28f69bb Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 10 years ago
  41. d6d5bff Initial GN work for WebRTC by kjellander@webrtc.org · 10 years ago
  42. e78505f Restore ptypes.txt file by henrik.lundin@webrtc.org · 10 years ago
  43. 38a2d46 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
  44. ab22857 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
  45. f0cf127 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
  46. 5c28a0a Update makefiles after merge of Chromium at 277521 by Android Chromium Automerger · 10 years ago
  47. e5a0f26 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af by Android Chromium Automerger · 10 years ago
  48. cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
  49. c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
  50. eddcc63 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  51. 847dfa5 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  52. e82b71d Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  53. d3a2886 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 10 years ago
  54. d998689 Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 10 years ago
  55. c1a2a43 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  56. f89ce46 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  57. 0c14539 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  58. d05de74 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
  59. bd98cef Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  60. 9257c64 Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
  61. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  62. dd32ef8 Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
  63. 555f957 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
  64. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  65. 0e43e6f Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  66. 5fcef2b Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 10 years ago
  67. 4150d6e Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 10 years ago
  68. f6eaabf Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  69. 4121fcd Revert 6405 "Update generated asm offsets scripts." by henrike@webrtc.org · 10 years ago
  70. 88417a9 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  71. 6298c29 Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  72. 3acaa1f Reland: Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  73. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  74. 3fe0d4f Revert 6395 "Making WebRTC able to play and record audio to file..." by minyue@webrtc.org · 10 years ago
  75. 994f778 Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  76. 5fe1bf7 Merge from Chromium at DEPS revision 36.0.1985.65 by Torne (Richard Coles) · 10 years ago
  77. 12914f8 Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 2efb7af01e3cbe87cb05d49893922e5f88a32115 by Torne (Richard Coles) · 10 years ago
  78. caf328c Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  79. ae4a452 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  80. 6e6b951 common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix by bjornv@webrtc.org · 10 years ago
  81. b96d9c7 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 10 years ago
  82. adda09e Update makefiles after merge of Chromium at 276202 by Android Chromium Automerger · 10 years ago
  83. 604ba6f Delete last file in neteq4 folder by henrik.lundin@webrtc.org · 10 years ago
  84. bc9c195 MIPS optimizations for ISAC (patch #1) by andrew@webrtc.org · 10 years ago
  85. d70e23e Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 10 years ago
  86. 9cd8281 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  87. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  88. bdcde22 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at bdfcddf7091e92134143e9a2d9ccce908e43979e by Android Chromium Automerger · 10 years ago
  89. eaaba5a Create a joint encoder/decoder wrapper for iSAC in ACM by henrik.lundin@webrtc.org · 10 years ago
  90. 6da16b3 Add thread annotations to AcmReceiver by henrik.lundin@webrtc.org · 10 years ago
  91. 8097a46 Update makefiles after merge of Chromium at 275833 by Android Chromium Automerger · 10 years ago
  92. b999e11 Make some methods in Clock class const declared by henrik.lundin@webrtc.org · 10 years ago
  93. 2efb7af Merge 6178 and 6273 from trunk to 3.53 (chrome 36) branch get the fix for windows NTP clock. by wu@webrtc.org · 10 years ago
  94. c05ab94 Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
  95. 688977c Merge from Chromium at DEPS revision 275586 by Torne (Richard Coles) · 10 years ago
  96. 377e7fd Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
  97. f03a4a6 Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
  98. 6d7c6e6 Multi-threaded unit test for Audio Coding Module using iSAC by henrik.lundin@webrtc.org · 10 years ago
  99. 4b50adf audio_processing: Forces extended filter to be used in splitting filter test. by bjornv@webrtc.org · 10 years ago
  100. e5abc85 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago