1 files changed, 23 insertions, 3 deletions
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index d86bfb553c..d50a53cb63 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -28,10 +28,12 @@
 #include "talk/media/webrtc/fakewebrtccall.h"
 
 #include <algorithm>
+#include <utility>
 
 #include "talk/media/base/rtputils.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/audio/audio_sink.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
@@ -39,14 +41,27 @@ FakeAudioSendStream::FakeAudioSendStream(
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
+const webrtc::AudioSendStream::Config&
+    FakeAudioSendStream::GetConfig() const {
+  return config_;
+}
+
 void FakeAudioSendStream::SetStats(
     const webrtc::AudioSendStream::Stats& stats) {
   stats_ = stats;
 }
 
-const webrtc::AudioSendStream::Config&
-    FakeAudioSendStream::GetConfig() const {
-  return config_;
+FakeAudioSendStream::TelephoneEvent
+    FakeAudioSendStream::GetLatestTelephoneEvent() const {
+  return latest_telephone_event_;
+}
+
+bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
+                                             uint32_t duration_ms) {
+  latest_telephone_event_.payload_type = payload_type;
+  latest_telephone_event_.event_code = event;
+  latest_telephone_event_.duration_ms = duration_ms;
+  return true;
 }
 
 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
@@ -77,6 +92,11 @@ webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
   return stats_;
 }
 
+void FakeAudioReceiveStream::SetSink(
+    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+  sink_ = std::move(sink);
+}
+
 FakeVideoSendStream::FakeVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const webrtc::VideoEncoderConfig& encoder_config)