Index: talk/app/webrtc/rtpsenderreceiver_unittest.cc |
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
index c9871864bd5618548c89efbdbb354c455868355f..3f61504cae36509b4c20fe3803e4c6f1354dec18 100644 |
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
@@ -26,10 +26,12 @@ |
*/ |
#include <string> |
+#include <utility> |
#include "talk/app/webrtc/audiotrack.h" |
#include "talk/app/webrtc/mediastream.h" |
#include "talk/app/webrtc/remoteaudiosource.h" |
+#include "talk/app/webrtc/remoteaudiotrack.h" |
#include "talk/app/webrtc/rtpreceiver.h" |
#include "talk/app/webrtc/rtpsender.h" |
#include "talk/app/webrtc/streamcollection.h" |
@@ -57,7 +59,8 @@ namespace webrtc { |
// Helper class to test RtpSender/RtpReceiver. |
class MockAudioProvider : public AudioProviderInterface { |
public: |
- virtual ~MockAudioProvider() {} |
+ ~MockAudioProvider() override {} |
+ |
MOCK_METHOD2(SetAudioPlayout, |
void(uint32_t ssrc, |
bool enable)); |
@@ -67,6 +70,14 @@ class MockAudioProvider : public AudioProviderInterface { |
const cricket::AudioOptions& options, |
cricket::AudioRenderer* renderer)); |
MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume)); |
+ |
+ void SetRawAudioSink(uint32_t, |
+ rtc::scoped_ptr<AudioSinkInterface> sink) override { |
+ sink_ = std::move(sink); |
+ } |
+ |
+ private: |
+ rtc::scoped_ptr<AudioSinkInterface> sink_; |
}; |
// Helper class to test RtpSender/RtpReceiver. |
@@ -151,8 +162,8 @@ class RtpSenderReceiverTest : public testing::Test { |
} |
void CreateAudioRtpReceiver() { |
- audio_track_ = |
- AudioTrack::Create(kAudioTrackId, RemoteAudioSource::Create().get()); |
+ audio_track_ = RemoteAudioTrack::Create( |
+ kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL)); |
EXPECT_TRUE(stream_->AddTrack(audio_track_)); |
EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true)); |
audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0], |