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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / f7795df4261877af9eaf982cccfafc3fc52aeb4a / voice_engine
  1. 7ea3607 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  2. bb1e3ff Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  3. 8773fa6 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  4. b0295bf Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  5. 3a87cff Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  6. 28e9b66 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  7. 5c6f3fd Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  8. 8ec46c6 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  9. ba9daa7 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. by henrika@webrtc.org · 10 years ago
  10. 616cbcd Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  11. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  12. 0061d86 * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  13. 110a2d2 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  14. 2e24460 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
  15. 8b4811b Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  16. a19bee3 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  17. a61127d Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  18. 13f9d37 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
  19. cf526f7 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  20. 514abde Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  21. 3ea24b2 Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  22. 14c5e8a Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 10 years ago
  23. 4f9c08f Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 10 years ago
  24. 1857d7e Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  25. a738ae3 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
  26. 0b559b6 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago
  27. 6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  28. 290c5a5 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  29. e338cc2 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
  30. 7a06daa (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
  31. eb90479 Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 10 years ago
  32. 4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  33. bf4f232 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
  34. c55faad Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
  35. 0725df6 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
  36. 5ae01bf Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
  37. 8f5ab19 VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
  38. 7c3f468 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
  39. 0027f0a Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
  40. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  41. d327be4 Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 10 years ago
  42. 40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  43. 55f4fe8 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  44. 4d9df07 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. by henrika@webrtc.org · 10 years ago
  45. 48bbc5a Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 10 years ago
  46. 5ddb6fe Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 10 years ago
  47. 8a4a39c Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  48. fa28e37 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  49. 8513671 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 10 years ago
  50. c8529ab Remove obsolete voe_unit_test. by solenberg@webrtc.org · 10 years ago
  51. ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  52. 0fd5775 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
  53. 48a5cdb Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
  54. 247df83 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 10 years ago
  55. dd1d6ce Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
  56. 1eba384 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  57. 4f41016 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  58. 0a7d406 Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  59. 910910a Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  60. 25bec2a Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
  61. 4f23307 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 10 years ago
  62. ccee3c3 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  63. 90c6679 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  64. 457e101 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  65. 842d07a Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  66. 926e88a Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  67. 9a7cb02 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  68. a3ae4d1 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  69. acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  70. b8dc2e2 Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  71. b50a841 Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  72. 7f0519e Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  73. 2f70422 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
  74. 894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  75. c8bd975 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  76. 5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  77. 36fb531 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  78. 4bfa866 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  79. 8dda8d2 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  80. e9274ae 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
  81. 7606f43 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  82. 2ba95be Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  83. 6c0739e 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
  84. 224c0f5 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  85. 1e6493d Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  86. 0f281aa Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  87. 9670be6 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  88. 28ea6f8 Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  89. 3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  90. 1364cf1 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  91. a064105 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  92. d24ce00 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  93. 0580c2c Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  94. bee99b1 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  95. e30fde1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  96. 1963a68 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  97. 3965d1f OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  98. d1deeb6 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  99. 81c4d24 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  100. 5632a64 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago