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Unified Diff: talk/media/webrtc/fakewebrtccall.h

Issue 1551813002: Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding unit tests for SetRawAudioSink. Created 4 years, 11 months ago
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Index: talk/media/webrtc/fakewebrtccall.h
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 3528c7a7b1e1e12a2290e71a5e9f5660bd5707f9..d6fe6871472e0c9d885142cbcd83dbdaa23a9e05 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -89,6 +89,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
int received_packets() const { return received_packets_; }
void IncrementReceivedPackets();
+ rtc::scoped_refptr<webrtc::AudioSinkInterface> sink() const { return sink_; }
private:
// webrtc::ReceiveStream implementation.
@@ -106,12 +107,12 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
- void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+ void SetSink(rtc::scoped_refptr<webrtc::AudioSinkInterface> sink) override;
webrtc::AudioReceiveStream::Config config_;
webrtc::AudioReceiveStream::Stats stats_;
int received_packets_;
- rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
+ rtc::scoped_refptr<webrtc::AudioSinkInterface> sink_;
};
class FakeVideoSendStream final : public webrtc::VideoSendStream,

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