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

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add thread check, Remove bits_per_sample and use int16_t. Created 5 years 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 024c50dd194fa10b6a7a3150b41dcfe1c114b942..3528c7a7b1e1e12a2290e71a5e9f5660bd5707f9 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -106,10 +106,12 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
+ void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
webrtc::AudioReceiveStream::Config config_;
webrtc::AudioReceiveStream::Stats stats_;
int received_packets_;
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
the sun 2015/12/11 16:32:04 again, I'd rather not add any extra gunk unless it
tommi (sloooow) - chröme 2015/12/11 17:51:18 Already used in existing tests. Before, there was
};
class FakeVideoSendStream final : public webrtc::VideoSendStream,

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