1 files changed, 47 insertions, 0 deletions
diff --git a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h
new file mode 100644
index 0000000..cd005d5
--- /dev/null
+++ b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_config.h
@@ -0,0 +1,47 @@
+// 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.
+
+#ifndef CAST_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_
+#define CAST_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_
+
+#include "media/cast/cast_config.h"
+#include "media/cast/rtp_common/rtp_defines.h"
+
+namespace media {
+namespace cast {
+
+struct RtpPacketizerConfig {
+  RtpPacketizerConfig() {
+    ssrc = 0;
+    max_payload_length = kIpPacketSize - 28;   // Default is IP-v4/UDP.
+    audio = false;
+    frequency = 8000;
+    payload_type = -1;
+    sequence_number = 0;
+    rtp_timestamp = 0;
+  }
+
+  // General.
+  bool audio;
+  int payload_type;
+  uint16 max_payload_length;
+  uint16 sequence_number;
+  uint32 rtp_timestamp;
+  int frequency;
+
+  // SSRC.
+  unsigned int ssrc;
+
+  // Video.
+  VideoCodec video_codec;
+
+  // Audio.
+  uint8 channels;
+  AudioCodec audio_codec;
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // CAST_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_