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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0d03514a1a95cd83126ba35102c4277555603577 / voice_engine
  1. 05384c0 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  2. 6b15b51 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  3. a84ddcd Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  4. 87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  5. c13a537 Move out typing detection to its own class. by henrikg@webrtc.org · 11 years ago
  6. 942ba53 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 11 years ago
  7. 0584274 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  8. b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  9. 22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  10. f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  11. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  12. 1a6b274 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  13. 5d7992f Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  14. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  15. 7d7e63d Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  16. f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  17. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  18. 27f0841 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  19. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  20. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  21. c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  22. 7950b98 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  23. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  24. b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  25. 7e97e4c Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  26. 76dad96 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  27. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  28. b27e670 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. by wu@webrtc.org · 11 years ago
  29. f7651ef Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  30. 3d553d4 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  31. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  32. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  33. e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  34. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  35. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  36. 8da2f65 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  37. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  38. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  39. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  40. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  41. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  42. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  43. 7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  44. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  45. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  46. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  47. 0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
  48. 0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  49. 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
  50. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  51. 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
  52. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  53. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  54. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  55. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  56. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  57. 1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  58. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  59. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  60. d171544 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 11 years ago
  61. 7d82c9d Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  62. a4a1afa Delete Channels without ChannelManager lock. by pbos@webrtc.org · 11 years ago
  63. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  64. f3bae63 Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  65. 44634a6 Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  66. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  67. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  68. 5ce8723 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  69. 0e6fa8c Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  70. 44f1239 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  71. 6349e17 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
  72. 1c8d5a0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  73. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  74. 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
  75. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  76. 6a4acb9 Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
  77. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  78. 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
  79. d5e5863 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  80. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  81. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  82. f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  83. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  84. 0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  85. 9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  86. 5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  87. 4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  88. b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  89. 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  90. a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  91. 92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  92. 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  93. 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
  94. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  95. 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  96. 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
  97. 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  98. 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  99. ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  100. 28832e1 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago