Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/stream_synchronization.h b/video_engine/stream_synchronization.h
new file mode 100644
index 0000000..25a370c
--- /dev/null
+++ b/video_engine/stream_synchronization.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
+#define WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
+
+#include <list>
+
+#include "modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
+#include "typedefs.h" // NOLINT
+
+namespace webrtc {
+
+struct ViESyncDelay;
+
+class StreamSynchronization {
+ public:
+ struct Measurements {
+ Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
+ synchronization::RtcpList rtcp;
+ int64_t latest_receive_time_ms;
+ uint32_t latest_timestamp;
+ };
+
+ StreamSynchronization(int audio_channel_id, int video_channel_id);
+ ~StreamSynchronization();
+
+ bool ComputeDelays(int relative_delay_ms,
+ int current_audio_delay_ms,
+ int* extra_audio_delay_ms,
+ int* total_video_delay_target_ms);
+
+ // On success |relative_delay| contains the number of milliseconds later video
+ // is rendered relative audio. If audio is played back later than video a
+ // |relative_delay| will be negative.
+ static bool ComputeRelativeDelay(const Measurements& audio_measurement,
+ const Measurements& video_measurement,
+ int* relative_delay_ms);
+
+ private:
+ ViESyncDelay* channel_delay_;
+ int audio_channel_id_;
+ int video_channel_id_;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_