Separating internal and external methods of RtpSender/RtpReceiver.

webrtc/api/rtpsender.cc

 /*
  *  Copyright 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 127 matching lines @@
     if (stats_) {
       stats_->AddLocalAudioTrack(track_.get(), ssrc_);
     }
   } else if (prev_can_send_track) {
     cricket::AudioOptions options;
     provider_->SetAudioSend(ssrc_, false, options, nullptr);
   }
   return true;
 }
 
+RtpParameters AudioRtpSender::GetParameters() const {
+  return provider_->GetAudioRtpSendParameters(ssrc_);
+}
+
+bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
+  return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
+}
+
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     cricket::AudioOptions options;
     provider_->SetAudioSend(ssrc_, false, options, nullptr);
     if (stats_) {
@@ skipping 42 matching lines @@
     // a remote audio track to a peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
 #endif
 
   cricket::AudioSource* source = sink_adapter_.get();
   ASSERT(source != nullptr);
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
 }
 
-RtpParameters AudioRtpSender::GetParameters() const {
-  return provider_->GetAudioRtpSendParameters(ssrc_);
-}
-
-bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
-  return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
-}
-
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                VideoProviderInterface* provider)
     : id_(track->id()),
       stream_id_(stream_id),
       provider_(provider),
       track_(track),
       cached_track_enabled_(track->enabled()) {
   RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
@@ skipping 61 matching lines @@
 
   // Update video provider.
   if (can_send_track()) {
     SetVideoSend();
   } else if (prev_can_send_track) {
     ClearVideoSend();
   }
   return true;
 }
 
+RtpParameters VideoRtpSender::GetParameters() const {
+  return provider_->GetVideoRtpSendParameters(ssrc_);
+}
+
+bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
+  return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
+}
+
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     ClearVideoSend();
   }
   ssrc_ = ssrc;
@@ skipping 27 matching lines @@
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), &options, track_);
 }
 
 void VideoRtpSender::ClearVideoSend() {
   RTC_DCHECK(ssrc_ != 0);
   RTC_DCHECK(provider_ != nullptr);
   provider_->SetVideoSend(ssrc_, false, nullptr, nullptr);
 }
 
-RtpParameters VideoRtpSender::GetParameters() const {
-  return provider_->GetVideoRtpSendParameters(ssrc_);
-}
-
-bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
-  return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
-}
-
 }  // namespace webrtc