Commit Graph

60 Commits

Author SHA1 Message Date
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
ccd42840bc Wire up statistics in video send stream of new video engine api
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@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 09:54:34 +00:00
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
724947b8ef Add SwapFrame() to VideoSendStreamInput.
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.

Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.

BUG=2657
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
4070935f4f Implement and test EncodedImageCallback in new ViE API.
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@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-26 11:41:59 +00:00
331d4402fc Connect pacer/padding to SuspendBelowMinBitrate
The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.

Had to change the unit test to make it aware of the padding packets.

BUG=2606
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 14:05:40 +00:00
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
ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate
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@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 12:18:43 +00:00
6488761f2e Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.

This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14 08:58:14 +00:00
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