Revert of Merge webrtc/video_engine/ into webrtc/video/

webrtc/video/payload_router.cc

Index: webrtc/video/payload_router.cc
diff --git a/webrtc/video/payload_router.cc b/webrtc/video/payload_router.cc
deleted file mode 100644
index 177f2dd4e85853b9458c1a31d9520f4323d39ef8..0000000000000000000000000000000000000000
--- a/webrtc/video/payload_router.cc
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- *  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