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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / vie_rtp_rtcp_impl.h
  1. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  2. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  3. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  4. 2f0c5f7 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
  5. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  6. 55a2a27 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  7. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  8. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  9. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  10. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  11. 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  12. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  13. 7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
  14. 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  15. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  16. 9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  17. 1619664 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  18. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago