commit 04f4931ef06273c2873e7816ed1f568d445117b8
author Fredrik Solenberg <solenberg@webrtc.org> Mon Jun 08 11:04:56 2015
committer Fredrik Solenberg <solenberg@webrtc.org> Mon Jun 08 11:05:07 2015
tree 319d43a4bf28c39dd7b3eadf9e0e1861e1de35ca
parent 77cabab51a2a362b589ba96ce81e605fbeca83e2

VoE2 API draft

BUG=4690
R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50029004

Cr-Commit-Position: refs/heads/master@{#9392}

webrtc/video/audio_receive_stream.cc

diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index de77f1b..f5383f4 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -63,6 +63,10 @@
   }
 }
 
+webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
+  return webrtc::AudioReceiveStream::Stats();
+}
+
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }

webrtc/video/audio_receive_stream.h

diff --git a/webrtc/video/audio_receive_stream.h b/webrtc/video/audio_receive_stream.h
index a321ec2..9935117 100644
--- a/webrtc/video/audio_receive_stream.h
+++ b/webrtc/video/audio_receive_stream.h
@@ -26,6 +26,8 @@
                      const webrtc::AudioReceiveStream::Config& config);
   ~AudioReceiveStream() override {}
 
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
   bool DeliverRtcp(const uint8_t* packet, size_t length);
   bool DeliverRtp(const uint8_t* packet, size_t length);
 

webrtc/video/call.cc

diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index bd96734..cde41bc 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -72,6 +72,10 @@
 
   PacketReceiver* Receiver() override;
 
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
@@ -196,6 +200,14 @@
 
 PacketReceiver* Call::Receiver() { return this; }
 
+webrtc::AudioSendStream* Call::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  return nullptr;
+}
+
+void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+}
+
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");