commit a52722fac401ae44fcc3fc67f9dd3055a18ea4ef
author eladalon <eladalon@webrtc.org> Mon Jun 26 18:23:54 2017
committer Commit Bot <commit-bot@chromium.org> Mon Jun 26 18:23:54 2017
tree 69cc9d24c09e376f1f3eb712fd6a142cf8862c89
parent 376b6fd483cc61171b5876156b0c675dba906b4f

Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )

Reason for revert:
About to fix problem and reland.

Original issue's description:
> Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> The problem is in the BUILD.gn file.
>
> Sample failure:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
>
> Sample logs:
> use_goma = true
> """ to /b/c/b/Linux_Builder/src/out/Release/args.gn.
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
>     "//webrtc/base:rtc_base_approved",
>     ^--------------------------------
>
> Original issue's description:
> > Create RtcpDemuxer. Capabilities:
> > 1. Demux RTCP messages according to the sender-SSRC.
> > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2943693003
> > Cr-Commit-Position: refs/heads/master@{#18763}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2957763002
> Cr-Commit-Position: refs/heads/master@{#18764}
> Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2960623002
Cr-Commit-Position: refs/heads/master@{#18768}

webrtc/call/rtp_packet_sink_interface.h

diff --git a/webrtc/call/rtp_packet_sink_interface.h b/webrtc/call/rtp_packet_sink_interface.h
index 900ca35..0b3e64e 100644
--- a/webrtc/call/rtp_packet_sink_interface.h
+++ b/webrtc/call/rtp_packet_sink_interface.h
@@ -14,10 +14,10 @@
 
 class RtpPacketReceived;
 
-// This class represents a receiver of an already parsed RTP packets.
+// This class represents a receiver of already parsed RTP packets.
 class RtpPacketSinkInterface {
  public:
-  virtual ~RtpPacketSinkInterface() {}
+  virtual ~RtpPacketSinkInterface() = default;
   virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
 };