CVO is, instead of rotating frame on the capture side, to have renderer rotate the frame based on a new rtp header extension.
The change includes
1. encoder side needs to pass this from raw frame to the encoded frame.
2. decoder needs to copy it from rtp packet (only the last packet of a frame has this info) to decoded frame.
R=mflodman@webrtc.orgTBR=stefan@webrtc.org
BUG=4145
Review URL: https://webrtc-codereview.appspot.com/46429006
Cr-Commit-Position: refs/heads/master@{#8767}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8767 4adac7df-926f-26a2-2b94-8c16560cd09d
> Speculative revert of 8631 "Remove lock from Bitrate() and FrameRate() in Video..."
>
> We ran into the alignment problem on Mac 10.9 debug again. This is the only CL I see in the range that adds an rtc::CriticalSection, so I'm trying out reverting it before attempting another roll.
>
> > Remove lock from Bitrate() and FrameRate() in VideoSender.
> > These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
> >
> > I'm making access to the rate variables from VCMGenericEncoder, thread safe, by using a lock that's not associated with the encoder. There should be little to no contention there. While modifying VCMGenericEncoder, I noticed that a couple of member variables weren't needed, so I removed them.
> >
> > The reason for this change is that getStats is currently contending with the encoder when Bitrate() is called. On my machine, this means that getStats can take about 25-30ms instead of ~1ms.
> >
> > Also adding some documentation for other methods and a suggestion for how we could avoid contention between the encoder and the network thread.
> >
> > BUG=2822
> > R=mflodman@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/43479004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/45529004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46519004
Cr-Commit-Position: refs/heads/master@{#8645}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8645 4adac7df-926f-26a2-2b94-8c16560cd09d
We ran into the alignment problem on Mac 10.9 debug again. This is the only CL I see in the range that adds an rtc::CriticalSection, so I'm trying out reverting it before attempting another roll.
> Remove lock from Bitrate() and FrameRate() in VideoSender.
> These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
>
> I'm making access to the rate variables from VCMGenericEncoder, thread safe, by using a lock that's not associated with the encoder. There should be little to no contention there. While modifying VCMGenericEncoder, I noticed that a couple of member variables weren't needed, so I removed them.
>
> The reason for this change is that getStats is currently contending with the encoder when Bitrate() is called. On my machine, this means that getStats can take about 25-30ms instead of ~1ms.
>
> Also adding some documentation for other methods and a suggestion for how we could avoid contention between the encoder and the network thread.
>
> BUG=2822
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/43479004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45529004
Cr-Commit-Position: refs/heads/master@{#8640}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8640 4adac7df-926f-26a2-2b94-8c16560cd09d
These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
I'm making access to the rate variables from VCMGenericEncoder, thread safe, by using a lock that's not associated with the encoder. There should be little to no contention there. While modifying VCMGenericEncoder, I noticed that a couple of member variables weren't needed, so I removed them.
The reason for this change is that getStats is currently contending with the encoder when Bitrate() is called. On my machine, this means that getStats can take about 25-30ms instead of ~1ms.
Also adding some documentation for other methods and a suggestion for how we could avoid contention between the encoder and the network thread.
BUG=2822
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43479004
Cr-Commit-Position: refs/heads/master@{#8631}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8631 4adac7df-926f-26a2-2b94-8c16560cd09d
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.
BUG=2346,2312
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2172004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
Also fixes a timer bug related to NACKing in the RTP module which could cause packets to only be NACKed twice if there's frequent packet losses.
Note that I decided to remove any selective NACKing for now as I don't think the gain of doing it is big enough compared to the added complexity. The same reasoning for empty packets. None of them will be retransmitted by a smart sender since the sender would know that they aren't needed.
BUG=1420
TEST=video_coding_unittests, vie_auto_test, video_coding_integrationtests, trybots
Review URL: https://webrtc-codereview.appspot.com/1115006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3599 4adac7df-926f-26a2-2b94-8c16560cd09d
