1 files changed, 79 insertions, 0 deletions
diff --git a/webrtc/video/vie_remb.h b/webrtc/video/vie_remb.h
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+++ b/webrtc/video/vie_remb.h
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+/*
+ *  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_VIE_REMB_H_
+#define WEBRTC_VIDEO_VIE_REMB_H_
+
+#include <list>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class ProcessThread;
+class RtpRtcp;
+
+class VieRemb : public RemoteBitrateObserver {
+ public:
+  explicit VieRemb(Clock* clock);
+  ~VieRemb();
+
+  // Called to add a receive channel to include in the REMB packet.
+  void AddReceiveChannel(RtpRtcp* rtp_rtcp);
+
+  // Removes the specified channel from REMB estimate.
+  void RemoveReceiveChannel(RtpRtcp* rtp_rtcp);
+
+  // Called to add a module that can generate and send REMB RTCP.
+  void AddRembSender(RtpRtcp* rtp_rtcp);
+
+  // Removes a REMB RTCP sender.
+  void RemoveRembSender(RtpRtcp* rtp_rtcp);
+
+  // Returns true if the instance is in use, false otherwise.
+  bool InUse() const;
+
+  // Called every time there is a new bitrate estimate for a receive channel
+  // group. This call will trigger a new RTCP REMB packet if the bitrate
+  // estimate has decreased or if no RTCP REMB packet has been sent for
+  // a certain time interval.
+  // Implements RtpReceiveBitrateUpdate.
+  virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
+                                       unsigned int bitrate);
+
+ private:
+  typedef std::list<RtpRtcp*> RtpModules;
+
+  Clock* const clock_;
+  rtc::scoped_ptr<CriticalSectionWrapper> list_crit_;
+
+  // The last time a REMB was sent.
+  int64_t last_remb_time_;
+  unsigned int last_send_bitrate_;
+
+  // All RtpRtcp modules to include in the REMB packet.
+  RtpModules receive_modules_;
+
+  // All modules that can send REMB RTCP.
+  RtpModules rtcp_sender_;
+
+  // The last bitrate update.
+  unsigned int bitrate_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIE_REMB_H_