Implementing a packet router class, used to route RTP packets to the
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.
BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39629004
Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video_engine/payload_router.h b/webrtc/video_engine/payload_router.h
new file mode 100644
index 0000000..3cc0bb0
--- /dev/null
+++ b/webrtc/video_engine/payload_router.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2015 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_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+#define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+
+#include <list>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RTPFragmentationHeader;
+class RtpRtcp;
+struct RTPVideoHeader;
+
+// PayloadRouter routes outgoing data to the correct sending RTP module, based
+// on the simulcast layer in RTPVideoHeader.
+class PayloadRouter {
+ public:
+ PayloadRouter();
+ ~PayloadRouter();
+
+ // Rtp modules are assumed to be sorted in simulcast index order.
+ void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
+
+ // PayloadRouter will only route packets if being active, all packets will be
+ // dropped otherwise.
+ void set_active(bool active);
+ bool active();
+
+ // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
+ // Returns true if the packet was routed / sent, false otherwise.
+ bool RoutePayload(FrameType frame_type,
+ int8_t payload_type,
+ uint32_t time_stamp,
+ int64_t capture_time_ms,
+ const uint8_t* payload_data,
+ size_t payload_size,
+ const RTPFragmentationHeader* fragmentation,
+ const RTPVideoHeader* rtp_video_hdr);
+
+ private:
+ scoped_ptr<CriticalSectionWrapper> crit_;
+
+ // Active sending RTP modules, in layer order.
+ std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
+ bool active_ GUARDED_BY(crit_.get());
+
+ DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_