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.cc b/webrtc/video_engine/payload_router.cc
new file mode 100644
index 0000000..f815d13
--- /dev/null
+++ b/webrtc/video_engine/payload_router.cc
@@ -0,0 +1,68 @@
+/*
+ * 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.
+ */
+
+#include "webrtc/video_engine/payload_router.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+
+namespace webrtc {
+
+PayloadRouter::PayloadRouter()
+ : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ active_(false) {}
+
+PayloadRouter::~PayloadRouter() {}
+
+void PayloadRouter::SetSendingRtpModules(
+ const std::list<RtpRtcp*>& rtp_modules) {
+ CriticalSectionScoped cs(crit_.get());
+ rtp_modules_.clear();
+ rtp_modules_.reserve(rtp_modules.size());
+ for (auto* rtp_module : rtp_modules) {
+ rtp_modules_.push_back(rtp_module);
+ }
+}
+
+void PayloadRouter::set_active(bool active) {
+ CriticalSectionScoped cs(crit_.get());
+ active_ = active;
+}
+
+bool PayloadRouter::active() {
+ CriticalSectionScoped cs(crit_.get());
+ return active_;
+}
+
+bool PayloadRouter::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) {
+ CriticalSectionScoped cs(crit_.get());
+ DCHECK(rtp_video_hdr == NULL ||
+ rtp_video_hdr->simulcastIdx <= rtp_modules_.size());
+
+ if (!active_ || rtp_modules_.empty())
+ return false;
+
+ int stream_idx = 0;
+ if (rtp_video_hdr != NULL)
+ stream_idx = rtp_video_hdr->simulcastIdx;
+ return rtp_modules_[stream_idx]->SendOutgoingData(
+ frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
+ payload_size, fragmentation, rtp_video_hdr) == 0 ? true : false;
+}
+
+} // namespace webrtc
