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64 Commits

Author SHA1 Message Date
pbos@webrtc.org
776e6f289c Use external VideoDecoders in VideoReceiveStream. 
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.

Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.

Additionally addresses a data race in VideoReceiver that was exposed with this change.

R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667

Review URL: https://webrtc-codereview.appspot.com/27829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 15:28:39 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:07:44 +00:00
pbos@webrtc.org
bd249bc711 Remove GetDefaultConfigs() from Call. 
Defaults for configs are instead placed in the Config constructors.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 04:45:15 +00:00
pbos@webrtc.org
a5c8d2c9b3 Rename Start/Stop in Video{Send,Receive}Streams. 
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.

R=mflodman@webrtc.org
BUG=3227

Review URL: https://webrtc-codereview.appspot.com/12329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:13:21 +00:00
sprang@webrtc.org
9510e53cc0 Make VideoReceiveStream::GetStats() const. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 15:32:45 +00:00
sprang@webrtc.org
09315705b9 Wire up statistics in video receive stream of new API 
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@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 12:06:29 +00:00
pbos@webrtc.org
c279a5d72c Wire up RTX in VideoReceiveStream. 
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.

BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 09:30:53 +00:00
pbos@webrtc.org
e02d47515f Set up receiver RTX config using a std::map. 
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 14:43:55 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00
mflodman@webrtc.org
b429e516a9 cpplint cleaning new API and its implementation files. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 09:46:22 +00:00
mflodman@webrtc.org
92c2793154 Adding REMB to receive stream configuration, the send side will always 
react to incoming REMB for now.

Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.

TEST=See above.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 16:36:28 +00:00
pbos@webrtc.org
b613b5ab2b Set local SSRC for VideoReceiveStream. 
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.

BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 10:13:04 +00:00
pbos@webrtc.org
53c8573525 Rename video streams' start/stop methods. 
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 11:36:47 +00:00
pbos@webrtc.org
16e03b7bd8 Separate Call API/build files from video_engine/. 
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@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 16:32:01 +00:00