Set the RTP payload type to fixed values for audio/video streams.

media/cast/net/rtp/rtp_sender.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/net/rtp/rtp_sender.h"
 
 #include "base/big_endian.h"
 #include "base/logging.h"
 #include "base/rand_util.h"
 #include "media/cast/constants.h"
@@ skipping 21 matching lines @@
       transport_task_runner_(transport_task_runner),
       weak_factory_(this) {
   // Randomly set sequence number start value.
   config_.sequence_number = base::RandInt(0, 65535);
 }
 
 RtpSender::~RtpSender() {}
 
 bool RtpSender::Initialize(const CastTransportRtpConfig& config) {
   config_.ssrc = config.ssrc;
-  config_.payload_type = static_cast<int>(config.rtp_payload_type);
+  // TODO(xjz): Android TV receivers expect the |payload_type| to be one of
+  // these two specific values. This constraint needs to be removed and the
+  // value of the |payload_type| can vary according to the spec:
+  // https://tools.ietf.org/html/rfc3551.
+  if (config.rtp_payload_type <= RtpPayloadType::AUDIO_LAST)
+    config_.payload_type = 127;
+  else
+    config_.payload_type = 96;
   packetizer_.reset(new RtpPacketizer(transport_, &storage_, config_));
   return true;
 }
 
 void RtpSender::SendFrame(const EncodedFrame& frame) {
   DCHECK(packetizer_);
   packetizer_->SendFrameAsPackets(frame);
   LOG_IF(DFATAL, storage_.GetNumberOfStoredFrames() > kMaxUnackedFrames)
       << "Possible bug: Frames are not being actively released from storage.";
 }
@@ skipping 99 matching lines @@
 int64_t RtpSender::GetLastByteSentForFrame(FrameId frame_id) {
   const SendPacketVector* stored_packets = storage_.GetFramePackets(frame_id);
   if (!stored_packets)
     return 0;
   PacketKey last_packet_key = stored_packets->rbegin()->first;
   return transport_->GetLastByteSentForPacket(last_packet_key);
 }
 
 }  //  namespace cast
 }  // namespace media