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

1 files changed, 91 insertions, 0 deletions

diff --git a/video_engine/vie_remb.h b/video_engine/vie_remb.h
new file mode 100644
index 00000000..43f373e0
--- /dev/null
+++ b/video_engine/vie_remb.h
@@ -0,0 +1,91 @@
+/*
+ *  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.
+ */
+
+// 1. Register a RtpRtcp module to include in the REMB packet.
+// 2. When UpdateBitrateEstimate is called for the first time for a SSRC, add it
+//    to the map.
+// 3. Send a new REMB every kRembSendIntervallMs or if a lower bitrate estimate
+//    for a specified SSRC.
+
+
+#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
+#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
+
+#include <list>
+#include <map>
+#include <utility>
+
+#include "modules/interface/module.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class ProcessThread;
+class RtpRtcp;
+
+class VieRemb : public RemoteBitrateObserver, public Module {
+ public:
+  explicit VieRemb(ProcessThread* process_thread);
+  ~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(unsigned int bitrate);
+
+  // Implements Module.
+  virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+  virtual WebRtc_Word32 TimeUntilNextProcess();
+  virtual WebRtc_Word32 Process();
+
+ private:
+  typedef std::list<RtpRtcp*> RtpModules;
+
+  ProcessThread* process_thread_;
+  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_;
+  int64_t bitrate_update_time_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_