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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / a89f7e8ccf950d6168ea55ca91ee892587eb6c91 / voice_engine
  1. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  2. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  3. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  4. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  5. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  6. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  7. 7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  8. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  9. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  10. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  11. 0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
  12. 0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  13. c766a74 Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. by dwkang@webrtc.org · 11 years ago
  14. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  15. e21b64b Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests). by henrike@webrtc.org · 11 years ago
  16. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  17. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  18. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  19. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  20. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  21. 1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  22. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  23. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  24. d171544 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 11 years ago
  25. 7d82c9d Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  26. a4a1afa Delete Channels without ChannelManager lock. by pbos@webrtc.org · 11 years ago
  27. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  28. f3bae63 Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  29. 44634a6 Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  30. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  31. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  32. 5ce8723 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  33. 0e6fa8c Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  34. 44f1239 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  35. 6349e17 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
  36. 1c8d5a0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  37. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  38. 9d788a1 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" by elham@webrtc.org · 11 years ago
  39. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  40. 6a4acb9 Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
  41. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  42. 446ea2e Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. by stefan@webrtc.org · 11 years ago
  43. d5e5863 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  44. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  45. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  46. f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  47. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  48. 0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  49. 9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  50. 5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  51. 4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  52. b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  53. 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  54. a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  55. 92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  56. 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  57. 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
  58. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  59. 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  60. d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  61. 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  62. 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  63. ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  64. 28832e1 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
  65. c0fc487 Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
  66. ad9cee8 Relax VoE's max packet length threshold. by andrew@webrtc.org · 11 years ago
  67. 9e0d9a6 Disabled flaky test. by phoglund@webrtc.org · 11 years ago
  68. 4a68e95 Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
  69. 166153e Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
  70. b6fadb1 Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 11 years ago
  71. 570c4a5 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  72. 237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
  73. f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  74. 54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  75. 8ec8955 Remove the old unused udp_transport by pwestin@webrtc.org · 11 years ago
  76. 1d25eac Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  77. ef91cbf Remove WEBRTC_*_ENGINE_NETWORK_API use by pwestin@webrtc.org · 11 years ago
  78. c39749a Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  79. 84423e9 Disabling MixingTests due to race conditions. by henrika@webrtc.org · 11 years ago
  80. 45ce6a8 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC by henrika@webrtc.org · 11 years ago
  81. e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  82. 9e8a401 Fixes memory leak in AudioLevel class reported by memory try bots. by henrika@webrtc.org · 11 years ago
  83. c4efe71 Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer by henrika@webrtc.org · 11 years ago
  84. 3b6f728 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  85. fa2dd22 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
  86. 2ffc8bf Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  87. 365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  88. f386e2b Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
  89. 31b4448 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  90. 13f66d1 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  91. 0c1f10b Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
  92. aa922de Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  93. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  94. 2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  95. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  96. 0f919be Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
  97. 8665399 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
  98. b79627b Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
  99. 4de0a10 Don't upsample the capture signal early. by andrew@webrtc.org · 11 years ago
  100. b563e5e Properly error check calls to AudioProcessing. by andrew@webrtc.org · 12 years ago