Move RTP for synchroninzation and rename classes, files and variables.
This CL removes (almost) the last RTP references in VideoReceiveStream.
There are still references to RTPFragmentationHeader and SSRCs, which
will be dealt with later.
There are also new GUARDED_BY and thred checker added to the
synchronization class.
When there are othre transports than RTP, there will instead be an
interface + inheritance for RtpStreamReceiver and
RtpStreamSynchronizattion in VideoReceiveStream. This work will be done
when we actually know how we want to make thee transport interface.
BUG=webrtc:5838
Review-Url: https://codereview.webrtc.org/2216533002
Cr-Commit-Position: refs/heads/master@{#13655}
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 1f116bf..84cab26 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -25,6 +25,8 @@
"report_block_stats.h",
"rtp_stream_receiver.cc",
"rtp_stream_receiver.h",
+ "rtp_streams_synchronizer.cc",
+ "rtp_streams_synchronizer.h",
"send_delay_stats.cc",
"send_delay_stats.h",
"send_statistics_proxy.cc",
@@ -47,8 +49,6 @@
"vie_encoder.h",
"vie_remb.cc",
"vie_remb.h",
- "vie_sync_module.cc",
- "vie_sync_module.h",
]
configs += [ "..:common_config" ]
