Adding getParameters/setParameters APIs to RtpReceiver.

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 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 1201 matching lines @@
 
 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
                                     std::unique_ptr<AudioSinkInterface> sink) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_)
     return;
 
   voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
 }
 
-RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetAudioRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (voice_channel_) {
-    return voice_channel_->GetRtpParameters(ssrc);
+    return voice_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetAudioRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetAudioRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     return false;
   }
-  return voice_channel_->SetRtpParameters(ssrc, parameters);
+  return voice_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetAudioRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetAudioRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool WebRtcSession::SetSource(
     uint32_t ssrc,
     rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
   ASSERT(signaling_thread()->IsCurrent());
 
   if (!video_channel_) {
     // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
     // support video.
@@ skipping 30 matching lines @@
     return;
   }
   if (!video_channel_->SetVideoSend(ssrc, enable, options)) {
     // Allow that MuteStream fail if |enable| is false but assert otherwise.
     // This in the normal case when the underlying media channel has already
     // been deleted.
     ASSERT(enable == false);
   }
 }
 
-RtpParameters WebRtcSession::GetVideoRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (video_channel_) {
-    return video_channel_->GetRtpParameters(ssrc);
+    return video_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetVideoRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetVideoRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!video_channel_) {
     return false;
   }
-  return video_channel_->SetRtpParameters(ssrc, parameters);
+  return video_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetVideoRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
     return false;
   }
   uint32_t send_ssrc = 0;
   // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
@@ skipping 816 matching lines @@
                                                   ssl_cipher_suite);
   }
 }
 
 void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc