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path: root/video_engine/vie_encoder.h
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2014-02-27Adds APIs for reporting pacer queuing delay.jiayl@webrtc.org
BUG=2775 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8959005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04Removing DropDeltaAfterKey functionality which is unused.andresp@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5214 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-26Implement and test EncodedImageCallback in new ViE API.sprang@webrtc.org
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Rename AutoMute to SuspendBelowMinBitratehenrik.lundin@webrtc.org
Changes all instances throughout the WebRTC stack. BUG=2436 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Fix for RTX in combination with pacing.stefan@webrtc.org
Retransmissions didn't get sent over RTX when pacing was enabled since the pacer didn't keep track of whether a packet was a retransmit or not. BUG=1811 TEST=trybots R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Separate Call API/build files from video_engine/.pbos@webrtc.org
BUG=2535 R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Removing the threshold from the auto-mute APIshenrik.lundin@webrtc.org
The threshold is now set equal to the minimum bitrate of the encoder. The test is also changed to have the REMB values depend on the minimum bitrate from the encoder. BUG=2436 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21Implement I420FrameCallbacks in Call.pbos@webrtc.org
BUG=2425 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2393004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18Have padding decay to zero if no frames are being captured.stefan@webrtc.org
BUG=1837 TEST=trybots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4998 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02Piping AutoMuter interface through to ViE APIhenrik.lundin@webrtc.org
This is a piece of the AutoMuter effort. A second CL will follow containing modifications to the new API, and tests. BUG=2436 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2331004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4899 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15Update talk to 50918584.wu@webrtc.org
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23Hooking up first simple CPU adaptation version.mflodman@webrtc.org
BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1767004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16Revert r4301tnakamura@webrtc.org
R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05Breaking out receive-stats, rtp-payload-registry and rtp-receiver from thestefan@webrtc.org
rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26Removed ViE file API.mflodman@webrtc.org
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1723004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20Enqueue packet in pacer if sending failshclam@chromium.org
If a packet cannot be sent while pacer is in use it should be queued. This avoid packet loss due to congestion. BUG=1930 R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1693004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Wire up pacer-based padding.stefan@webrtc.org
This connects the pacer-based padding with the RTP modules, which will generate padding packets roughly according to what the pacer suggests. It will only generate padding packets of maximum size to keep the number off padding packets as small as possible. This also sets a limit of how much padding + media bitrate which the pacer is allowed to "request" from the RTP modules. Padding will for now only be generated by the first sending RTP module. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1612005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17Include files from webrtc/.. paths in video_engine/pbos@webrtc.org
BUG=1662 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1492004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13Wiring down config from video engine until video coding and remote bitrate ↵andresp@webrtc.org
estimator modules instantiation. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1450008 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02Adding trace and changing pacing constantspwestin@webrtc.org
BUG=1721,1722 R=mikhal@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1380005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09WebRtc_Word32 -> int32_t in video_engine/pbos@webrtc.org
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1302005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-27Add interface to signal a network down event.stefan@webrtc.org
- In real-time mode encoding will be paused until the network is back up. - In buffering mode the encoder will keep encoding, and packets will be buffered at the sender. When the buffer grows above the target delay encoding will be paused. - Fixes a couple of issues related to pacing which was found with the new test. - Introduces different max bitrates for pacing and for encoding. This allows the pacer to faster get rid of the queue after a network down event. (Work based on issue 1237004) BUG=1524 TESTS=trybots,vie_auto_test Review URL: https://webrtc-codereview.appspot.com/1258004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15Adding a receive side API for buffering mode.mikhal@webrtc.org
At the same time, renaming the send side API. Review URL: https://webrtc-codereview.appspot.com/1104004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-10Updates to send side streaming mode:mikhal@webrtc.org
1. Disabling frame-droppers from the vie encoder and not the channel. 2. Accounting for qpMax in the VP8 wrapper. Review URL: https://webrtc-codereview.appspot.com/1101007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09Enable external encoders with internal picture source.stefan@webrtc.org
CL enables registering of external encoder with internal picture source on API by adding simple passthrough parameter that is already supported within video engine. Review URL: https://webrtc-codereview.appspot.com/1006006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13Enable paced sender. pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/965016 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25Only remove encoder state feedback for send channels.mflodman@webrtc.org
BUG=1000 TEST=See bug Review URL: https://webrtc-codereview.appspot.com/938004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2994 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25Revert the revert in r2988 since that wasn't the issue.mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/931005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24Reverse Merged r2884 & r2888 from trunk.vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24Switching to I420VideoFramemikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/922004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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