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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / voice_engine
  1. 658bfc4 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b031016337f09d758a97ed51b67788e574431103 by Android Chromium Automerger · 10 years ago
  2. 945d969 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  3. 467f756 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
  4. 6ce3720 Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  5. e064ae4 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  6. eae2214 Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  7. ff164b8 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
  8. 708ff4d Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  9. 692224a 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
  10. 66ccaff 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
  11. 1b5b5c5 Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 10 years ago
  12. bee17f6 Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 10 years ago
  13. 1a07e42 Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  14. 8fb9156 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
  15. bbea098 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago
  16. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  17. 4ff0eda 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
  18. 91d88e1 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
  19. 1307f73 Update makefiles after merge of Chromium at 263101 by Android Chromium Automerger · 10 years ago
  20. 54b48fa Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 55bc2810c06fe624311518c4502af5ca8a5c085c by Android Chromium Automerger · 10 years ago
  21. c71dd0d Update makefiles after merge of Chromium at 262754 by Android Chromium Automerger · 10 years ago
  22. 2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
  23. 01f4592 Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 10 years ago
  24. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  25. 97a64e2 Update makefiles after merge of Chromium at 262110 by Android Chromium Automerger · 10 years ago
  26. e111a22 Update makefiles after merge of Chromium at 261622 by Android Chromium Automerger · 10 years ago
  27. f7c73b5 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
  28. 965e921 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at cceb392c510580dbdf1cec2eec17e9c9ccb268ab by Android Chromium Automerger · 10 years ago
  29. cceb392 Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
  30. 120c725 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
  31. 56aeb0e Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
  32. 2ebb6ba Update makefiles after merge of Chromium at 260927 by Android Chromium Automerger · 10 years ago
  33. 172f42a VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
  34. c9dca91 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
  35. 2bf87a2 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
  36. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  37. 8d49b3f Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 10 years ago
  38. 18c2945 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  39. a5db8e3 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  40. 650772a 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
  41. df08c5d Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 10 years ago
  42. 3f7753c Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 10 years ago
  43. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  44. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  45. ad065d0 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 10 years ago
  46. f6c4fc3 Remove obsolete voe_unit_test. by solenberg@webrtc.org · 10 years ago
  47. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  48. 9a95ac2 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 10 years ago
  49. 4c20478 Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 10 years ago
  50. 22c954a Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 10 years ago
  51. 8ad10ec Restore mixing integration tests. by andrew@webrtc.org · 10 years ago
  52. 05384c0 Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" by pbos@webrtc.org · 10 years ago
  53. 6b15b51 Fix locking in LoopBackTransport::StorePacket. by pbos@webrtc.org · 10 years ago
  54. a84ddcd Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents by marpan@webrtc.org · 10 years ago
  55. 87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
  56. c13a537 Move out typing detection to its own class. by henrikg@webrtc.org · 10 years ago
  57. 942ba53 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
  58. 0584274 Output logs to stderr from voe_cmd_test by default. by andrew@webrtc.org · 11 years ago
  59. b3ff385 Temporarily disabling some more audio processing tests. by aluebs@webrtc.org · 11 years ago
  60. 22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
  61. f3a2ef3 Android: Fixes crash when exiting WebRTCDemo. by henrike@webrtc.org · 11 years ago
  62. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  63. 1a6b274 Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  64. 5d7992f Fix the include guard in transmit_mixer.h by braveyao@webrtc.org · 11 years ago
  65. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  66. 7d7e63d Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  67. f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  68. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  69. 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
  70. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  71. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  72. c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  73. 7950b98 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  74. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  75. b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  76. 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
  77. 76dad96 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  78. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  79. 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
  80. f7651ef Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  81. 3d553d4 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  82. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  83. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  84. e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  85. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  86. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  87. 8da2f65 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  88. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  89. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  90. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  91. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  92. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  93. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  94. 7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  95. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  96. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  97. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  98. 0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
  99. 0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  100. 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