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}
diff --git a/webrtc/call/rtcp_packet_sink_interface.h b/webrtc/call/rtcp_packet_sink_interface.h
new file mode 100644
index 0000000..e26bd37
--- /dev/null
+++ b/webrtc/call/rtcp_packet_sink_interface.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
+#define WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
+
+#include "webrtc/base/array_view.h"
+
+namespace webrtc {
+
+// This class represents a receiver of unparsed RTCP packets.
+// TODO(eladalon): Replace this by demuxing over parsed rather than raw data.
+// Whether this should be over an entire RTCP packet, or over RTCP blocks,
+// is still under discussion.
+class RtcpPacketSinkInterface {
+ public:
+ virtual ~RtcpPacketSinkInterface() = default;
+ virtual void OnRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
