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gerrit-public.fairphone.software / platform / external / webrtc / b935d489914099f33e2e474b022978dfab91f301 / pc / rtp_sender.cc
  1. 619b294 RtpSender's RtpParameters were invalidated in a call to SLD/SRD. by Amit Hilbuch · 6 years ago
  2. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 6 years ago
  3. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  4. aa58415 Reland "Enabling Simulcast use via AddTransceiver." by Amit Hilbuch · 6 years ago
  5. 7832343 Revert "Enabling Simulcast use via AddTransceiver." by Emircan Uysaler · 6 years ago
  6. ce470aa Enabling Simulcast use via AddTransceiver. by Amit Hilbuch · 6 years ago
  7. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 years ago
  8. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  9. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/rtpsender.cc]
  10. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  11. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  12. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  13. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  14. 95ca6e1 AudioSource allows implementations to return settings by Piotr (Peter) Slatala · 6 years ago
  15. c462a6e Prevent the frame decryptor being set if the channel is stopped. by Benjamin Wright · 6 years ago
  16. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  17. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 6 years ago
  18. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  19. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  20. 8c1bf95 Reland "Add initial support for RtpEncodingParameters max_framerate." by Åsa Persson · 6 years ago
  21. bfd412e Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. by Benjamin Wright · 6 years ago
  22. d81ac95 Injects FrameEncryptorInterface into RtpSender. by Benjamin Wright · 6 years ago
  23. 948b7e3 Revert "Add initial support for RtpEncodingParameters max_framerate." by Mirko Bonadei · 6 years ago
  24. ced5cfd Add initial support for RtpEncodingParameters max_framerate. by Åsa Persson · 6 years ago
  25. 87b3c51 Implement changing degradation preference with setParameters() by Florent Castelli · 6 years ago
  26. 111fdfd Refactor RtpSender to take the sender ID as a constructor argument by Steve Anton · 7 years ago
  27. c19ab07 Add support for content-hint value "text" by Harald Alvestrand · 7 years ago
  28. b983bae Remove unused/deprecated DTMF methods by Steve Anton · 7 years ago
  29. 5565981 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. by Åsa Persson · 7 years ago
  30. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  31. abe301f Add HeaderExtensions to RtpParameters by Florent Castelli · 7 years ago
  32. 2d2c888 Returns RTCError for setting unimplemented RtpParameters. by Seth Hampson · 7 years ago
  33. cebf50f Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Florent Castelli · 7 years ago
  34. 909338b Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Max Morin · 7 years ago
  35. 5faf36e Implement RtpParameters.transaction_id for PC RtpSenderInterface by Florent Castelli · 7 years ago
  36. ff40b14 Delete obsolete enable argument to SetVideoSend. by Niels Möller · 7 years ago
  37. 5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 7 years ago
  38. 191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 7 years ago
  39. f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 7 years ago
  40. bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 7 years ago
  41. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 7 years ago
  42. 3d976f6 Discard link to media channel when audio sender stopped. by Harald Alvestrand · 7 years ago
  43. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  44. 45cc890 Assorted logging pedantry by Jonas Olsson · 7 years ago
  45. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  46. 2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
  47. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  48. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  49. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  50. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  51. 02ee47c Signal track ID correctly when Unified Plan semantics selected by Steve Anton · 7 years ago
  52. f9381f0 Implement PeerConnection::AddTrack/RemoveTrack for Unified Plan by Steve Anton · 7 years ago
  53. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  54. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  55. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  56. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  57. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsender.cc]
  58. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  59. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  60. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  61. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  62. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  63. b11fb25 Protect APM in webkit builds. by agouaillard · 8 years ago
  64. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  65. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/rtpsender.cc]
  66. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  67. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  68. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  69. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  70. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  71. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  72. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  73. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  74. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago
  75. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  76. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  77. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 9 years ago
  78. dabc944 Add missing tracing to RtpSender objects. by Peter Boström · 9 years ago
  79. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  80. c0d31e9 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 9 years ago
  81. 9e083d2 Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ ) by perkj · 9 years ago
  82. 246b527 Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ ) by deadbeef · 9 years ago
  83. c9022f5 Delete empty API files and cleaned up includes. by perkj · 9 years ago
  84. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  85. 0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 9 years ago
  86. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  87. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
  88. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  89. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/rtpsender.cc]
  90. 6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
  91. 3c16978 Remove cast to LocalAudioSource from AudioRtpSender. by Tommi · 9 years ago
  92. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  93. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  94. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
  95. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  96. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  97. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  98. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  99. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  100. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago