Commit Graph

7 Commits

Author SHA1 Message Date
737336d37a Add NACK rate throttling for audio channels.
Not really used for audio today (already in place for video), but should
still function anyway.

BUG=
R=henrika@webrtc.org, minyue@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2181383002 .

Cr-Commit-Position: refs/heads/master@{#13571}
2016-07-29 10:59:49 +00:00
ec4f068bcd Style cleanups in RtpSender.
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h

R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2067673004 .

Cr-Commit-Position: refs/heads/master@{#13565}
2016-07-28 22:19:18 +00:00
0208322ee3 GN: Add video_engine_tests
Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 10:53:09 +00:00
733b5478dd Movable support for VideoReceiveStream::Config and avoid copies.
Instead of the default copy constructor, the Copy() method has to be used.  In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream.  Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case).  Most importantly, creating copies is made harder and the interface encourages ownership transfers.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2042603002 .

Cr-Commit-Position: refs/heads/master@{#13102}
2016-06-10 15:58:12 +00:00
dc7d0d2ef0 Move, almost, all receive side references to RTP to RtpStreamReceiver.
There are still a few places in VideoReceiveStream where the RTP module
is explicitly used, e.g. setting up a/v sync, but it's a bigger task to
change and that will be done in a follow up instead of in this CL.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/1947913002
Cr-Commit-Position: refs/heads/master@{#12642}
2016-05-06 12:32:30 +00:00
cfc8e3b9ef Removed all RTP dependencies from ViEChannel and renamed class.
ViEChannel is now called VideoStreamReceiver.

There will be a follow up CL removing all rtp references from VideoReceiveStream, but that made this CL to big and it will be done separately.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/1929313002
Cr-Commit-Position: refs/heads/master@{#12619}
2016-05-04 04:22:12 +00:00
fa66659c6b Rename ViEReceiver and move ownership to VideoReceiveStream.
This CL will be followed up with another CL removing everything related
to RTP from ViEChannel to RtpStreamReceiver, i.e. remove ViEChannel::rtp_stream_receiver_.

BUG=5838

Review-Url: https://codereview.webrtc.org/1917363005
Cr-Commit-Position: refs/heads/master@{#12553}
2016-04-29 06:15:38 +00:00