1 files changed, 74 insertions, 0 deletions
diff --git a/chromium/media/cast/rtp_sender/rtp_sender.h b/chromium/media/cast/rtp_sender/rtp_sender.h
new file mode 100644
index 00000000000..f6e59acba84
--- /dev/null
+++ b/chromium/media/cast/rtp_sender/rtp_sender.h
@@ -0,0 +1,74 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// This file contains the interface to the cast RTP sender.
+
+#ifndef MEDIA_CAST_RTP_SENDER_RTP_SENDER_H_
+#define MEDIA_CAST_RTP_SENDER_RTP_SENDER_H_
+
+#include <map>
+#include <set>
+
+#include "base/memory/scoped_ptr.h"
+#include "base/time/default_tick_clock.h"
+#include "base/time/tick_clock.h"
+#include "base/time/time.h"
+#include "media/cast/cast_config.h"
+#include "media/cast/rtp_sender/packet_storage/packet_storage.h"
+#include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.h"
+#include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h"
+
+namespace media {
+namespace cast {
+
+class PacedPacketSender;
+struct RtcpSenderInfo;
+
+typedef std::map<uint8, std::set<uint16> > MissingFramesAndPackets;
+
+// This object is only called from the main cast thread.
+// This class handles splitting encoded audio and video frames into packets and
+// add an RTP header to each packet. The sent packets are stored until they are
+// acknowledged by the remote peer or timed out.
+class RtpSender {
+ public:
+  RtpSender(const AudioSenderConfig* audio_config,
+            const VideoSenderConfig* video_config,
+            PacedPacketSender* transport);
+
+  ~RtpSender();
+
+  // The video_frame objects ownership is handled by the main cast thread.
+  void IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame,
+                                 const base::TimeTicks& capture_time);
+
+  // The audio_frame objects ownership is handled by the main cast thread.
+  void IncomingEncodedAudioFrame(const EncodedAudioFrame* audio_frame,
+                                 const base::TimeTicks& recorded_time);
+
+  void ResendPackets(const MissingFramesAndPackets& missing_packets);
+
+  void RtpStatistics(const base::TimeTicks& now, RtcpSenderInfo* sender_info);
+
+  // Used for testing.
+  void set_clock(base::TickClock* clock) {
+    // TODO(pwestin): review how we pass in a clock for testing.
+    clock_ = clock;
+  }
+
+ private:
+  void UpdateSequenceNumber(std::vector<uint8>* packet);
+
+  RtpPacketizerConfig config_;
+  scoped_ptr<RtpPacketizer> packetizer_;
+  scoped_ptr<PacketStorage> storage_;
+  PacedPacketSender* transport_;
+  scoped_ptr<base::TickClock> default_tick_clock_;
+  base::TickClock* clock_;
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_RTP_SENDER_RTP_SENDER_H_