Return a copy of the supported RTP header extensions instead of a reference.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 499 matching lines @@
 
   signal_thread_checker_.DetachFromThread();
   std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
 
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
   webrtc::Trace::SetTraceCallback(this);
 
   // Load our audio codec list.
   codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
 
-  // Load our RTP Header extensions.
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
-                         kRtpAudioLevelHeaderExtensionDefaultId));
-  rtp_header_extensions_.push_back(
-      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
-                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
-  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
-    rtp_header_extensions_.push_back(RtpHeaderExtension(
-        kRtpTransportSequenceNumberHeaderExtension,
-        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
-  }
   options_ = GetDefaultEngineOptions();
   voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
 }
 
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
   if (adm_) {
     voe_wrapper_.reset();
     adm_->Release();
@@ skipping 526 matching lines @@
   unsigned int ulevel;
   return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
       static_cast<int>(ulevel) : -1;
 }
 
 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   return codecs_;
 }
 
-const std::vector<RtpHeaderExtension>&
-WebRtcVoiceEngine::rtp_header_extensions() const {
+RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return rtp_header_extensions_;
+  RtpCapabilities capabilities;
+  capabilities.header_extensions.push_back(RtpHeaderExtension(
+      kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
+  capabilities.header_extensions.push_back(
+      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
+    capabilities.header_extensions.push_back(RtpHeaderExtension(
+        kRtpTransportSequenceNumberHeaderExtension,
+        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
+  }
+  return capabilities;
 }
 
 int WebRtcVoiceEngine::GetLastEngineError() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->error();
 }
 
 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
                               int length) {
   // Note: This callback can happen on any thread!
@@ skipping 1546 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE