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      1 /*
      2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #include "webrtc/video/payload_router.h"
     12 
     13 #include "webrtc/base/checks.h"
     14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
     15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
     16 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
     17 
     18 namespace webrtc {
     19 
     20 PayloadRouter::PayloadRouter()
     21     : crit_(CriticalSectionWrapper::CreateCriticalSection()),
     22       active_(false) {}
     23 
     24 PayloadRouter::~PayloadRouter() {}
     25 
     26 size_t PayloadRouter::DefaultMaxPayloadLength() {
     27   const size_t kIpUdpSrtpLength = 44;
     28   return IP_PACKET_SIZE - kIpUdpSrtpLength;
     29 }
     30 
     31 void PayloadRouter::SetSendingRtpModules(
     32     const std::list<RtpRtcp*>& rtp_modules) {
     33   CriticalSectionScoped cs(crit_.get());
     34   rtp_modules_.clear();
     35   rtp_modules_.reserve(rtp_modules.size());
     36   for (auto* rtp_module : rtp_modules) {
     37     rtp_modules_.push_back(rtp_module);
     38   }
     39 }
     40 
     41 void PayloadRouter::set_active(bool active) {
     42   CriticalSectionScoped cs(crit_.get());
     43   active_ = active;
     44 }
     45 
     46 bool PayloadRouter::active() {
     47   CriticalSectionScoped cs(crit_.get());
     48   return active_ && !rtp_modules_.empty();
     49 }
     50 
     51 bool PayloadRouter::RoutePayload(FrameType frame_type,
     52                                  int8_t payload_type,
     53                                  uint32_t time_stamp,
     54                                  int64_t capture_time_ms,
     55                                  const uint8_t* payload_data,
     56                                  size_t payload_length,
     57                                  const RTPFragmentationHeader* fragmentation,
     58                                  const RTPVideoHeader* rtp_video_hdr) {
     59   CriticalSectionScoped cs(crit_.get());
     60   if (!active_ || rtp_modules_.empty())
     61     return false;
     62 
     63   // The simulcast index might actually be larger than the number of modules in
     64   // case the encoder was processing a frame during a codec reconfig.
     65   if (rtp_video_hdr != NULL &&
     66       rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
     67     return false;
     68 
     69   int stream_idx = 0;
     70   if (rtp_video_hdr != NULL)
     71     stream_idx = rtp_video_hdr->simulcastIdx;
     72   return rtp_modules_[stream_idx]->SendOutgoingData(
     73       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
     74       payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
     75 }
     76 
     77 void PayloadRouter::SetTargetSendBitrates(
     78     const std::vector<uint32_t>& stream_bitrates) {
     79   CriticalSectionScoped cs(crit_.get());
     80   if (stream_bitrates.size() < rtp_modules_.size()) {
     81     // There can be a size mis-match during codec reconfiguration.
     82     return;
     83   }
     84   int idx = 0;
     85   for (auto* rtp_module : rtp_modules_) {
     86     rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
     87   }
     88 }
     89 
     90 size_t PayloadRouter::MaxPayloadLength() const {
     91   size_t min_payload_length = DefaultMaxPayloadLength();
     92   CriticalSectionScoped cs(crit_.get());
     93   for (auto* rtp_module : rtp_modules_) {
     94     size_t module_payload_length = rtp_module->MaxDataPayloadLength();
     95     if (module_payload_length < min_payload_length)
     96       min_payload_length = module_payload_length;
     97   }
     98   return min_payload_length;
     99 }
    100 
    101 }  // namespace webrtc
    102