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gerrit-public.fairphone.software / platform / external / webrtc / abe301fe6c5008f9a421e5794907685beec52c75 / media / engine / webrtcvoiceengine_unittest.cc
  1. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 6 years ago
  2. f859e55 Removing warning suppression flags from media. by Mirko Bonadei · 6 years ago
  3. dacec71 Add Rtcp parameters for PeerConnection senders by Florent Castelli · 6 years ago
  4. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 7 years ago
  5. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  6. 5a26a3a Remove public sync_label from StreamParams by Steve Anton · 7 years ago
  7. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  8. a1f6661 Check that channel is in "send" before OKing DTMF by Harald Alvestrand · 7 years ago
  9. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  10. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  11. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  12. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  13. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  14. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  15. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  16. fa266ef Fix the crash when GetSources is called with non-existing ssrc. by Zhi Huang · 7 years ago
  17. e26456a Removes usage of AGC APIs in the ADM. by henrika · 7 years ago
  18. 606a597 Remove adjust_agc_delta from WebRtcVoiceEngine by Steve Anton · 7 years ago
  19. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  20. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  21. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  22. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  23. 7880758 Optional: Use nullopt and implicit construction in /media by Oskar Sundbom · 7 years ago
  24. c97cf03 Removes unused sample-rate APIs from the ADM. by henrika · 7 years ago
  25. 8962b54 Removes Set/GetRecordingChannel() from the ADM by henrika · 7 years ago
  26. e78bcb9 Enable cpplint in media/ by Steve Anton · 7 years ago
  27. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  28. 919dc2e Removes fallback from Linux PulseAudio to ALSA. by henrika · 7 years ago
  29. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  30. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  31. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  32. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  33. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  34. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  35. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/engine/webrtcvoiceengine_unittest.cc]
  36. 0d0b912 Add and modify a few ANA stats. by ivoc · 7 years ago
  37. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  38. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  39. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  40. b1c9d1d Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine by peah · 7 years ago
  41. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  42. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  43. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  44. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  45. eb02c03 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 7 years ago
  46. 418b7d3 Increase number of unsignaled audio streams we handle to 4. by solenberg · 7 years ago
  47. 048cbdd Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ ) by aleloi · 7 years ago
  48. fe9ecb0 Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ ) by aleloi · 7 years ago
  49. d2303a2 Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ ) by aleloi · 7 years ago
  50. be68b72 Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ ) by aleloi · 7 years ago
  51. c61bf94 Activate 'offload debug dump recordings from audio thread to TaskQueue'. by aleloi · 7 years ago
  52. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  53. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  54. cb38367 Allow a received audio codec's payload type to change. by deadbeef · 7 years ago
  55. 3bc1510 Fix RtpReceiver.GetParameters when SSRCs aren't signaled. by deadbeef · 7 years ago
  56. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  57. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  58. 83862e3 Remove VoECodec from FakeWebRtcVoiceEngine. by solenberg · 8 years ago
  59. cecec10 Set max bitrate for audio send stream based on RtpParameters. by minyue · 8 years ago
  60. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  61. 1ccf73f Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video. by stefan · 8 years ago
  62. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  63. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  64. 9a5f03222 Remove VoEHardware interface usage. by solenberg · 8 years ago
  65. ebb349d Revert to allowing only 1 unsignaled receive stream for audio. by solenberg · 8 years ago
  66. 0c4b849 Pick a matching CN codec, rather than the first CN codec. by ossu · 8 years ago
  67. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  68. 2100c0b Support N unsignaled audio streams. by solenberg · 8 years ago
  69. 322a9e4 Handle TimeUntilNextProcess in StartupShutdownWithExternalADM by tommi · 8 years ago
  70. 76377c5 Remove usage of VoEAudioProcessing from WVoE/MC. by solenberg · 8 years ago
  71. 4904fb6 Be less pessimistic about turning "default" receive streams into signaled streams. by solenberg · 8 years ago
  72. 11bfc53 Fixed a couple of build-flag dependent tests of webrtcvoiceengine. by ossu · 8 years ago
  73. bcd88db WebRtcVoiceEngineTest: Changed a static_cast to a checked_cast. by ossu · 8 years ago
  74. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  75. 9def800 Added a flag to AudioCodecSpec to indicate adaptive bitrate support. by ossu · 8 years ago
  76. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  77. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  78. e1405ad Removed double-special-casing of ISAC in libjingle and WebRtcVoE. by ossu · 8 years ago
  79. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  80. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  81. 6672b26 Add overhead to audio bwe min, max. by michaelt · 8 years ago
  82. fb2aced Add video send SSRC to RtpParameters, and don't allow changing SSRC. by deadbeef · 8 years ago
  83. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  84. 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  85. f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  86. cb44343 Add SSRC to RtpEncodingParameters for audio. by deadbeef · 8 years ago
  87. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  88. 13f1a0a Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel. by stefan · 8 years ago
  89. 8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
  90. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  91. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  92. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  93. 2779bab Support receiving DTMF for multiple RTP clock rates. by solenberg · 8 years ago
  94. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  95. 3663c52 Provide move semantic for cricket::Codec and subclasses by magjed · 8 years ago
  96. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  97. 059fb44 - Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing. by solenberg · 8 years ago
  98. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  99. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  100. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago