1 files changed, 101 insertions, 0 deletions
diff --git a/webrtc/video/payload_router.cc b/webrtc/video/payload_router.cc
new file mode 100644
index 0000000000..177f2dd4e8
--- /dev/null
+++ b/webrtc/video/payload_router.cc
@@ -0,0 +1,101 @@
+/*
+ *  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/payload_router.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+PayloadRouter::PayloadRouter()
+    : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      active_(false) {}
+
+PayloadRouter::~PayloadRouter() {}
+
+size_t PayloadRouter::DefaultMaxPayloadLength() {
+  const size_t kIpUdpSrtpLength = 44;
+  return IP_PACKET_SIZE - kIpUdpSrtpLength;
+}
+
+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_ && !rtp_modules_.empty();
+}
+
+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_length,
+                                 const RTPFragmentationHeader* fragmentation,
+                                 const RTPVideoHeader* rtp_video_hdr) {
+  CriticalSectionScoped cs(crit_.get());
+  if (!active_ || rtp_modules_.empty())
+    return false;
+
+  // The simulcast index might actually be larger than the number of modules in
+  // case the encoder was processing a frame during a codec reconfig.
+  if (rtp_video_hdr != NULL &&
+      rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
+    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_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
+}
+
+void PayloadRouter::SetTargetSendBitrates(
+    const std::vector<uint32_t>& stream_bitrates) {
+  CriticalSectionScoped cs(crit_.get());
+  if (stream_bitrates.size() < rtp_modules_.size()) {
+    // There can be a size mis-match during codec reconfiguration.
+    return;
+  }
+  int idx = 0;
+  for (auto* rtp_module : rtp_modules_) {
+    rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
+  }
+}
+
+size_t PayloadRouter::MaxPayloadLength() const {
+  size_t min_payload_length = DefaultMaxPayloadLength();
+  CriticalSectionScoped cs(crit_.get());
+  for (auto* rtp_module : rtp_modules_) {
+    size_t module_payload_length = rtp_module->MaxDataPayloadLength();
+    if (module_payload_length < min_payload_length)
+      min_payload_length = module_payload_length;
+  }
+  return min_payload_length;
+}
+
+}  // namespace webrtc