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gerrit-public.fairphone.software / platform / external / webrtc / 01611ed3ac4102dc209e50bd618dd7933e11f314 / pc / rtp_sender.cc
  1. b05ca4b Implement new specification for degradation preference by Florent Castelli · 4 years, 6 months ago
  2. 3f1aee3 Change network_priority from a double to an enum. by Taylor Brandstetter · 4 years, 6 months ago
  3. e77912b Insert frame transformer between Encoded and Packetizer. by Marina Ciocea · 4 years, 6 months ago
  4. 99d6d81 Adding absolute capture timestamp to AudioTrackSinkInterface. by Minyue Li · 4 years, 7 months ago
  5. a8c2f51 Remove unused non-standard RtpEncodingParameters members by Florent Castelli · 4 years, 9 months ago
  6. cb11a31 Guard GenerateUniqueId() against concurrent access. by Yves Gerey · 5 years ago
  7. 1ff16c8 Add RtpSenderInterface.SetStreams by Guido Urdaneta · 5 years ago
  8. cc18917 Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface" by Henrik Andreassson · 5 years ago
  9. df5731e Improve spec compliance of SetStreamIDs in RtpSenderInterface by Guido Urdaneta · 5 years ago
  10. 619b294 RtpSender's RtpParameters were invalidated in a call to SLD/SRD. by Amit Hilbuch · 5 years ago
  11. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 6 years ago
  12. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  13. aa58415 Reland "Enabling Simulcast use via AddTransceiver." by Amit Hilbuch · 6 years ago
  14. 7832343 Revert "Enabling Simulcast use via AddTransceiver." by Emircan Uysaler · 6 years ago
  15. ce470aa Enabling Simulcast use via AddTransceiver. by Amit Hilbuch · 6 years ago
  16. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 years ago
  17. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  18. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/rtpsender.cc]
  19. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  20. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  21. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  22. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  23. 95ca6e1 AudioSource allows implementations to return settings by Piotr (Peter) Slatala · 6 years ago
  24. c462a6e Prevent the frame decryptor being set if the channel is stopped. by Benjamin Wright · 6 years ago
  25. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  26. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 6 years ago
  27. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  28. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  29. 8c1bf95 Reland "Add initial support for RtpEncodingParameters max_framerate." by Åsa Persson · 6 years ago
  30. bfd412e Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. by Benjamin Wright · 6 years ago
  31. d81ac95 Injects FrameEncryptorInterface into RtpSender. by Benjamin Wright · 6 years ago
  32. 948b7e3 Revert "Add initial support for RtpEncodingParameters max_framerate." by Mirko Bonadei · 6 years ago
  33. ced5cfd Add initial support for RtpEncodingParameters max_framerate. by Åsa Persson · 6 years ago
  34. 87b3c51 Implement changing degradation preference with setParameters() by Florent Castelli · 6 years ago
  35. 111fdfd Refactor RtpSender to take the sender ID as a constructor argument by Steve Anton · 6 years ago
  36. c19ab07 Add support for content-hint value "text" by Harald Alvestrand · 6 years ago
  37. b983bae Remove unused/deprecated DTMF methods by Steve Anton · 6 years ago
  38. 5565981 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. by Åsa Persson · 6 years ago
  39. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  40. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 6 years ago
  41. 2d2c888 Returns RTCError for setting unimplemented RtpParameters. by Seth Hampson · 6 years ago
  42. cebf50f Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Florent Castelli · 6 years ago
  43. 909338b Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Max Morin · 6 years ago
  44. 5faf36e Implement RtpParameters.transaction_id for PC RtpSenderInterface by Florent Castelli · 6 years ago
  45. ff40b14 Delete obsolete enable argument to SetVideoSend. by Niels Möller · 6 years ago
  46. 5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 6 years ago
  47. 191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 6 years ago
  48. f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 6 years ago
  49. bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 6 years ago
  50. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 6 years ago
  51. 3d976f6 Discard link to media channel when audio sender stopped. by Harald Alvestrand · 6 years ago
  52. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 6 years ago
  53. 45cc890 Assorted logging pedantry by Jonas Olsson · 7 years ago
  54. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  55. 2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
  56. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  57. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  58. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  59. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  60. 02ee47c Signal track ID correctly when Unified Plan semantics selected by Steve Anton · 7 years ago
  61. f9381f0 Implement PeerConnection::AddTrack/RemoveTrack for Unified Plan by Steve Anton · 7 years ago
  62. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  63. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  64. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  65. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  66. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsender.cc]
  67. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  68. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  69. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  70. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  71. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  72. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  73. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  74. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/rtpsender.cc]
  75. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  76. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  77. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  78. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  79. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  80. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  81. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  82. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  83. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
  84. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  85. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  86. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 8 years ago
  87. dabc944 Add missing tracing to RtpSender objects. by Peter Boström · 8 years ago
  88. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  89. c0d31e9 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 8 years ago
  90. 9e083d2 Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ ) by perkj · 8 years ago
  91. 246b527 Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ ) by deadbeef · 8 years ago
  92. c9022f5 Delete empty API files and cleaned up includes. by perkj · 8 years ago
  93. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  94. 0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 8 years ago
  95. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  96. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 8 years ago
  97. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  98. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/rtpsender.cc]
  99. 6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
  100. 3c16978 Remove cast to LocalAudioSource from AudioRtpSender. by Tommi · 9 years ago