Simplify SetFecParameters signature.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 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 1118 matching lines @@
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
   assert(!audio_configured_ && "Sender is an audio stream!");
   return video_->VideoCodecType();
 }
 
 void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
   RTC_DCHECK(!audio_configured_);
   video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
 }
 
-int32_t RTPSender::SetFecParameters(
-    const FecProtectionParams *delta_params,
-    const FecProtectionParams *key_params) {
+bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
+                                 const FecProtectionParams& key_params) {
   if (audio_configured_) {
-    return -1;
+    return false;
   }
   video_->SetFecParameters(delta_params, key_params);
-  return 0;
+  return true;
 }
 
 std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
     const RtpPacketToSend& packet) {
   // TODO(danilchap): Create rtx packet with extra capacity for SRTP
   // when transport interface would be updated to take buffer class.
   std::unique_ptr<RtpPacketToSend> rtx_packet(new RtpPacketToSend(
       &rtp_header_extension_map_, packet.size() + kRtxHeaderSize));
   // Add original RTP header.
   rtx_packet->CopyHeaderFrom(packet);
@@ skipping 81 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc