Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / voice_engine / channel.cc
  1. b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
  2. 237d079 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. by stefan@webrtc.org · 10 years ago
  3. 1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 10 years ago
  4. b9ca3e2 Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 10 years ago
  5. ca4bc68 Remove timestamp retreival warning/error. by turaj@webrtc.org · 10 years ago
  6. 31b38da Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. by minyue@webrtc.org · 10 years ago
  7. 9fbd3ec Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  8. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  9. 81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
  10. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  11. dd671de This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. by minyue@webrtc.org · 10 years ago
  12. 91c0a25 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED by minyue@webrtc.org · 10 years ago
  13. 881a32d Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  14. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  15. 7d20dda Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  16. 7f5e297 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
  17. 12884ba 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
  18. 3cd0f7c Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  19. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  20. 47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  21. 6ce3720 Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  22. e064ae4 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  23. eae2214 Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  24. 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
  25. 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
  26. 1a07e42 Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  27. 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
  28. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  29. f7c73b5 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
  30. c9dca91 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
  31. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  32. a5db8e3 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  33. df08c5d Resolves TSan v2 warnings in voe_auto_test. by henrika@webrtc.org · 10 years ago
  34. 3f7753c Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. by braveyao@webrtc.org · 10 years ago
  35. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  36. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  37. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  38. e95dc25 Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 11 years ago
  39. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  40. f1b92fd Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  41. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  42. 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
  43. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  44. 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
  45. f7651ef Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  46. e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  47. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  48. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  49. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  50. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  51. 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
  52. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  53. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  54. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  55. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  56. 5ce8723 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  57. 0e6fa8c Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  58. 44f1239 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  59. 1c8d5a0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  60. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  61. 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
  62. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  63. 6a4acb9 Fix some voe_auto_test uninitialised-value errors. by pbos@webrtc.org · 11 years ago
  64. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  65. 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
  66. d5e5863 Initialize payload-type frequency in channel.cc. by pbos@webrtc.org · 11 years ago
  67. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  68. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  69. 9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  70. 5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  71. 4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  72. b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  73. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  74. 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
  75. ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  76. 570c4a5 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
  77. f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  78. 54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  79. 1d25eac Resolves TSan v2 reports data races in voe_auto_test. by henrika@webrtc.org · 11 years ago
  80. e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  81. 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
  82. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  83. 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
  84. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  85. 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
  86. e9bb4e5 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
  87. 39ac132 Revert 3543 by tina.legrand@webrtc.org · 11 years ago
  88. 01ee1ba Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
  89. ead8a5b Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
  90. 2344ebe Fix propagating RED paylaod-type to ACM. by turaj@webrtc.org · 11 years ago
  91. 040f800 fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). by turaj@webrtc.org · 11 years ago
  92. 7547680 Fixes payload spelling error. by henrike@webrtc.org · 12 years ago
  93. d468236 Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 12 years ago
  94. db32ab0 Make VoE handle longer delays by niklas.enbom@webrtc.org · 12 years ago
  95. b9e3afc Add GetAudioFrame API to VoiceEngine. by roosa@google.com · 12 years ago
  96. 90d333e Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 12 years ago
  97. ca77149 Add API to retreive last received RTP timestamp to VoiceEngine. by roosa@google.com · 12 years ago
  98. 7db5290 VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
  99. c65ae4b Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
  100. d6f028b VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago