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webrtc / src / webrtc / cb436f6ea8999c8fde4c83e94be275dcb72ae0cd / audio
  1. b9f62aa Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  2. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  3. af8f6e5 Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  4. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  5. d6bc12c Fix two invalid DCHECKs related to audio BWE. by stefan · 8 years ago
  6. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  7. 085ed30 Support multiple connected Android devices in low bandwidth audio test by oprypin · 8 years ago
  8. 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  9. 7843bda Add Windows, Mac, Android support to low bandwidth audio test by oprypin · 8 years ago
  10. a54ea6e Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 8 years ago
  11. f67ccf5 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 8 years ago
  12. 9b9b7e2 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  13. e153ae8 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
  14. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  15. 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  16. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  17. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  18. 1993d91 Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  19. 49d1987 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  20. 1b0415d Low-bandwidth audio testing by oprypin · 8 years ago
  21. a58b581 Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ ) by nisse · 8 years ago
  22. 7a8eb23 Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ ) by skvlad · 8 years ago
  23. b1d9258 Delete class MockCongestionController. by nisse · 8 years ago
  24. 4e519ca Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  25. 109f6f7 Add low_bandwidth_audio_test to default build by oprypin · 8 years ago
  26. 9268abe Move delay_based_bwe_ into CongestionController by elad.alon · 8 years ago
  27. 0f20d58 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) by oprypin · 8 years ago
  28. 899c143 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) by oprypin · 8 years ago
  29. ba76094 Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 8 years ago
  30. 68e2137 Adding placeholder for low bandwidth audio test by kjellander · 8 years ago
  31. 5d07023 Remove MockRemoteBitrateObserver (unused) by elad.alon · 8 years ago
  32. bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  33. f21613d Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
  34. 9617a87 Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  35. 95a13b4 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  36. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
  37. 660a2e0 Clean out platform specific things from voice_engine_defines.h. by solenberg · 8 years ago
  38. c10c31d Wire up audio packet loss to BWE. by stefan · 8 years ago
  39. 49cdd35 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  40. f65fb4c Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  41. 6045993 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  42. a0add70 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  43. 5cacd3d Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  44. 40ab430 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  45. 9551ae0 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  46. 6a60706 Enable periodic bitrate probing when application limited for audio BWE. by stefan · 8 years ago
  47. 90cc2a9 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
  48. df5ec64 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
  49. 5c59d8f Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  50. bc2b639 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  51. 3e52797 Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  52. ee5a316 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  53. ff6e0cd GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries. by ehmaldonado · 8 years ago
  54. e10c50d Add ossu@ to OWNERS of audio/ and modules/audio_coding/ by ossu · 8 years ago
  55. 800a8f1 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  56. 818a3ee Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  57. 0808545 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  58. 27e08d9 Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
  59. 31212ba Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  60. d7ef04b Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  61. 2bc91bf Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  62. c01dfe6 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  63. 61e42da Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  64. 25b4c2e Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
  65. 13d9326 RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  66. 4cb29f6 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  67. acd8db6 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  68. 4d9c52e Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  69. 441ecb3 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  70. 9a1d49f Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  71. d85b51d Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  72. 7713f33 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  73. ac14ca2 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  74. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  75. abef9e9 Remove all references to GYP by Henrik Kjellander · 8 years ago
  76. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  77. 2c7aecc Make use of new APM statistics interface. by ivoc · 8 years ago
  78. a7f3617 Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago
  79. 2454551 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  80. a1912ef Fixing config for Audio BWE. by minyue · 8 years ago
  81. def0180 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 8 years ago
  82. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  83. 5206602 Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  84. b87bc5d Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  85. 91c6f34 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  86. 77b3ea6 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  87. 5928a3d Fix BWE simulations so that it uses the delay based BWE. by terelius · 8 years ago
  88. b8bff80 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  89. dfb640f Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  90. fbd0246 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  91. b4ac6b0 Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  92. bffc190 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  93. a6c6623 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  94. 5e0ea59 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  95. 6dcc486 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  96. d0dcb3b Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  97. e59b6ff Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  98. 36a2479 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 9 years ago
  99. 15b966d Enable the -Wundef warning for clang by kwiberg · 9 years ago
  100. 2cf29b5 GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 9 years ago