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path: root/common_types.h
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2014-07-11Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.stefan@webrtc.org
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems. R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Count total bytes sent in RTPSender::Bytes().pbos@webrtc.org
Previously only media bytes were included, this adds header bytes and padding bytes to the calculation. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
R=mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17849005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15Add ToString() to VideoSendStream::Config.pbos@webrtc.org
Adds ToString() to subsequent parts as well as a common.gyp to define ToString() methods for config.h. VideoStream is also moved to config.h. BUG=3171 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29Made common_types.h PacketTime declaration match ↵henrike@webrtc.org
https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 BUG=N/A R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6020 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17Removes parts of the VoEBase sub API as part of a clean-up operation where ↵henrika@webrtc.org
the goal is to remove unused APIs. BUG=3206 R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12019005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10Remove usage of webrtc trace in video processing modules.asapersson@webrtc.org
BUG=3153 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11089005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25Removing VideoCodecDerived and moving methods inside VideoCodec.mallinath@webrtc.org
VideoCodecDerived added to handle changes to talk (fakewebrtcvideoengine.h). R=mflodman@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5784 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24Add targetBitrate to VideoCodec struct.pbos@webrtc.org
To be used by a codec implementation. Could for instance be interpreted as trying to fit as much as possible on one temporal layer and send everything that doesn't fit within target bitrate on another one. Prevents an existing hack where startBitrate is used by a codec implementation to signify target bitrate. This hack forces a reset of bitrate estimation to target bitrate which creates bitrate dips. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5759 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-21Adding operator== and != methods for CodecInst and VideoCodec structures.mallinath@webrtc.org
R=juberti@google.com, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10099005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Remove internal codecs from VideoSendStream.pbos@webrtc.org
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings struct. The EncoderSettings struct uses an external encoder for all codecs. This means that external users, such as libjingle, will provide the encoders themselves, removing the previous distinction of internal and external codecs. For now VideoSendStream translates to VideoCodec internally. In the interrim (before the corresponding change is implemented in VideoReceiveStream) tests convert EncoderSettings to VideoCodecs. Removes Call::GetVideoCodecs(). Disables RampUpTest.WithPacingAndRtx as its further exposed with changes to bitrates used in tests. BUG=2854,2992 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19Add RTCP packet type counter (for getting statistics such as sent/received ↵asapersson@webrtc.org
NACK and FIR). Add counter to RTCP sender and RTCP receiver. Add video api GetRtcpPacketTypes(). BUG=2638 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
BUG= R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07Wire up statistics in video receive stream of new APIsprang@webrtc.org
This CL includes Call tests that test both send and receive sides. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14Remove empty VideoCodecGeneric struct.pbos@webrtc.org
Struct was added prematurely and triggers a warning with -Wextern-c-compat in latest clang. R=henrika@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/7119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07Wire up statistics in video send stream of new video engine apisprang@webrtc.org
Note, this CL does not contain any tests. Those are implemeted as call tests and will be submitted when the receive stream is wired up as well. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5559006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16Revert r5294 to re-roll r5293.pbos@webrtc.org
To fix races in test each stream now owns its own encoder/decoder. R=mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/5919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."turaj@webrtc.org
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. > > BUG= > R=mflodman@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5409004 TBR=solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Auto instantiate RBE depending on whether AST or TOF is available in ↵solenberg@webrtc.org
incoming packet stream. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Update talk to 58174641 together with http://review.webrtc.org/4319005/.wu@webrtc.org
R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Callback for send bitrate estimates - new rollsprang@webrtc.org
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as r5259, after which flakiness was detected and a rollback was performed at r5261. Patch Set 1 of this issue is the code submitted in r5259. Subsequent patch sets fixes a race condition which caused the seen problems. The root cause was a dead lock between a thread sending rtp packets and and a timed module processing thread: webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock webrtc::Bitrate::Process() // Get Bitrate lock webrtc::RTPSender::ProcessBitrate() webrtc::ModuleRtpRtcpImpl::Process() ... webrtc::Bitrate::Update() // Get Bitrate lock webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock webrtc::RTPSender::SendToNetwork() ... This is fixed in Bitrate::Process() by releasing the lock before calling the callback. BUG=2235 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Update talk to 58127566 together withwu@webrtc.org
https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Revert 5274 "Update talk to 58113193 together with https://webrt..."wu@webrtc.org
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. > > R=mallinath@webrtc.org, niklas.enbom@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5719004 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Update talk to 58113193 together with ↵wu@webrtc.org
https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11Revert 5259 "Callback for send bitrate estimates"sprang@webrtc.org
CL is causing flakiness in RampUpTest.WithoutPacing. > Callback for send bitrate estimates > > BUG=2235 > R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/4459004 R=mflodman@webrtc.org, pbos@webrtc.org TBR=mflodman Review URL: https://webrtc-codereview.appspot.com/5579005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11Callback for send bitrate estimatessprang@webrtc.org
BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05Add callbacks for send channel rtp statisticssprang@webrtc.org
BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05Add callbacks for send channel rtcp statisticssprang@webrtc.org
BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04Add send frame rate statistics callbacksprang@webrtc.org
BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21Make interface destructor virtualsprang@webrtc.org
In summary, do this: - ~FrameCountObserver() {} + virtual ~FrameCountObserver() {} BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4099005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5148 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Interface changes to old api, for use by new api transition.sprang@webrtc.org
BUG=2589 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.sprang@webrtc.org
We will do some refactoring of video engine and would like to use the same rtcp stats struct there. Both video and audio seem to use 8bit fraction lost, so that is changed in the struct as well. BUG= R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09Remove redundant STR_CASE_CMP macro definitions.andrew@webrtc.org
R=minyue@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2187005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23webrtc/common_types.h: Document bitrate fields' units.fischman@webrtc.org
R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1847004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4386 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14Replace ExtraCodecOptions with new Config class that supports multiple ↵andresp@webrtc.org
settings at once. R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1452004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4017 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."pbos@webrtc.org
> Revert 3933 "Remove traces of deprecated WebRtc_Word types." > > > Remove traces of deprecated WebRtc_Word types. > > > > BUG=314 > > R=tommi@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/1385004 > > TBR=pbos@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/1386004 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1397004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02Revert 3933 "Remove traces of deprecated WebRtc_Word types."pbos@webrtc.org
> Remove traces of deprecated WebRtc_Word types. > > BUG=314 > R=tommi@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/1385004 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1386004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02Remove traces of deprecated WebRtc_Word types.pbos@webrtc.org
BUG=314 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1385004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23Adding extra options to interact with external encoder/decoder.andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1327006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3893 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE ↵solenberg@webrtc.org
SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. Today I had to figure out this code was legacy. Now next person doesn't have to. BUG= Review URL: https://webrtc-codereview.appspot.com/1247004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22Add min and target bitrate to VideoCodec.marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1214004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3710 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-18Generic video-codec support.pbos@webrtc.org
Labels frames as key/delta, also marks the first RTP packet of a frame as such, to allow proper reconstruction even if packets are received out of order. BUG=1442 TBR=ajm@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1207004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12Remove DTMF detection. Talk team has been in the loop and there is no need forturaj@webrtc.org
DTMF detection at the receiver side. test=voe_auto_test, VoE extended test DTMF Review URL: https://webrtc-codereview.appspot.com/1168004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does ↵turaj@webrtc.org
not provide an API to change the mode of the background noise. bug=issue1370 test=trybots Review URL: https://webrtc-codereview.appspot.com/1121007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-18Refactoring temporal layers implementation and adding VideoCodecMode for ↵stefan@webrtc.org
easier control of codec settings. Review URL: https://webrtc-codereview.appspot.com/1105007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31VP8: Making key frame interval a tunnable parametermikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1070006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14Add number of inserted samples to NetEq statistics.roosa@google.com
BUG= Review URL: https://webrtc-codereview.appspot.com/964030 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12Expose NetEq playout mode off through VoiceEngine.roosa@google.com
BUG= Review URL: https://webrtc-codereview.appspot.com/971016 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-20Add a kTraceTerseInfo level for non-verbose logging.andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/937023 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3134 4adac7df-926f-26a2-2b94-8c16560cd09d
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