Delete unused method PayloadRouter::MaxPayloadLength.

webrtc/video/payload_router.cc

 /*
  *  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.
  */
 
@@ skipping 80 matching lines @@
 
 PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
                              int payload_type)
     : active_(false),
       rtp_modules_(rtp_modules),
       payload_type_(payload_type) {
 }
 
 PayloadRouter::~PayloadRouter() {}
 
-size_t PayloadRouter::DefaultMaxPayloadLength() {
-  const size_t kIpUdpSrtpLength = 44;
-  return IP_PACKET_SIZE - kIpUdpSrtpLength;
-}
-
 void PayloadRouter::SetActive(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)
     return;
   active_ = active;
 
   for (auto& module : rtp_modules_) {
     module->SetSendingStatus(active_);
     module->SetSendingMediaStatus(active_);
   }
@@ skipping 26 matching lines @@
   bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
       encoded_image._frameType, payload_type_, encoded_image._timeStamp,
       encoded_image.capture_time_ms_, encoded_image._buffer,
       encoded_image._length, fragmentation, &rtp_video_header, &frame_id);
   if (!send_result)
     return Result(Result::ERROR_SEND_FAILED);
 
   return Result(Result::OK, frame_id);
 }
 
-size_t PayloadRouter::MaxPayloadLength() const {
-  size_t min_payload_length = DefaultMaxPayloadLength();
-  rtc::CritScope lock(&crit_);
-  for (size_t i = 0; i < rtp_modules_.size(); ++i) {
-    size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
-    if (module_payload_length < min_payload_length)
-      min_payload_length = module_payload_length;
-  }
-  return min_payload_length;
-}
-
 void PayloadRouter::OnBitrateAllocationUpdated(
     const BitrateAllocation& bitrate) {
   rtc::CritScope lock(&crit_);
   if (IsActive()) {
     if (rtp_modules_.size() == 1) {
       // If spatial scalability is enabled, it is covered by a single stream.
       rtp_modules_[0]->SetVideoBitrateAllocation(bitrate);
     } else {
       // Simulcast is in use, split the BitrateAllocation into one struct per
       // rtp stream, moving over the temporal layer allocation.
       for (size_t si = 0; si < rtp_modules_.size(); ++si) {
         BitrateAllocation layer_bitrate;
         for (int tl = 0; tl < kMaxTemporalStreams; ++tl)
           layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl));
         rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate);
       }
     }
   }
 }
 
 }  // namespace webrtc