.:
[WebKit-https.git] / Source / WebCore / platform / mediastream / gstreamer / RealtimeOutgoingAudioSourceLibWebRTC.h
index ee56d52..53d15cb 100644 (file)
 
 #if USE(LIBWEBRTC)
 
+#include "GStreamerAudioStreamDescription.h"
+#include "GStreamerCommon.h"
 #include "RealtimeOutgoingAudioSource.h"
 
+#include <gst/audio/audio.h>
+
 namespace WebCore {
 
 class RealtimeOutgoingAudioSourceLibWebRTC final : public RealtimeOutgoingAudioSource {
@@ -43,6 +47,13 @@ private:
     bool hasBufferedEnoughData() final;
 
     void pullAudioData() final;
+
+    GstAudioConverter* m_sampleConverter;
+    std::unique_ptr<GStreamerAudioStreamDescription> m_inputStreamDescription;
+    std::unique_ptr<GStreamerAudioStreamDescription> m_outputStreamDescription;
+
+    Lock m_adapterMutex;
+    GRefPtr<GstAdapter> m_adapter;
 };
 
 } // namespace WebCore