Cast:Moving netwrok sender related code to a designated folder

media/cast/net/rtp_sender/rtp_packetizer/test/rtp_header_parser.h

Index: media/cast/net/rtp_sender/rtp_packetizer/test/rtp_header_parser.h
diff --git a/media/cast/net/rtp_sender/rtp_packetizer/test/rtp_header_parser.h b/media/cast/net/rtp_sender/rtp_packetizer/test/rtp_header_parser.h
new file mode 100644
index 0000000000000000000000000000000000000000..33e31fafc58f3fe2771d3787301c5d3a58e52cad
--- /dev/null
+++ b/media/cast/net/rtp_sender/rtp_packetizer/test/rtp_header_parser.h
@@ -0,0 +1,75 @@
+// 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.
+
+// Utility parser for rtp packetizer unittests
+#ifndef MEDIA_CAST_NET_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_
+#define MEDIA_CAST_NET_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_
+
+#include "base/basictypes.h"
+#include "media/cast/net/cast_net_defines.h"
+
+namespace media {
+namespace cast {
+
+struct RtpCastTestHeader {

[pwestin, 2013/12/10 23:03:01]

confused about this struct why do we need it?

[mikhal1, 2013/12/11 00:29:33]

Just for testing. Before I was using the RtpCastHeader which includes Webrtc.
This is now moved to the rtp_receiver, and therefore not available here.
On 2013/12/10 23:03:01, pwestin wrote:

+  RtpCastTestHeader() {
+    is_key_frame = false;
+    frame_id = 0;
+    packet_id = 0;
+    max_packet_id = 0;
+    is_reference = false;
+    reference_frame_id = 0;
+    marker = false;
+    sequence_number = 0;
+    rtp_timestamp = 0;
+    ssrc = 0;
+    payload_type = 0;
+    num_csrcs = 0;
+    audio_num_energy = 0;
+    header_length = 0;
+  }
+  //webrtc::WebRtcRTPHeader webrtc;
+  // Cast specific.
+  bool is_key_frame;
+  uint32 frame_id;
+  uint16 packet_id;
+  uint16 max_packet_id;
+  bool is_reference;  // Set to true if the previous frame is not available,
+                      // and the reference frame id  is available.
+  uint32 reference_frame_id;
+
+  // Rtp Generic.
+  bool marker;
+  uint16 sequence_number;
+  uint32 rtp_timestamp;
+  uint32 ssrc;
+  int payload_type;
+  uint8 num_csrcs;
+  uint8 audio_num_energy;
+  int header_length;
+
+};
+
+class RtpHeaderParser {
+ public:
+  RtpHeaderParser(const uint8* rtpData, size_t rtpDataLength);
+  ~RtpHeaderParser();
+
+  bool Parse(RtpCastTestHeader* parsed_packet) const;
+ private:

[pwestin, 2013/12/10 23:03:01]

line break

[mikhal1, 2013/12/11 00:29:33]

Done.

+  bool ParseCommon(RtpCastTestHeader* parsed_packet) const;
+  bool ParseCast(RtpCastTestHeader* parsed_packet) const;
+  const uint8* const rtp_data_begin_;
+  size_t length_;
+
+  mutable FrameIdWrapHelper frame_id_wrap_helper_;
+  mutable FrameIdWrapHelper reference_frame_id_wrap_helper_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpHeaderParser);
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_NET_RTP_SENDER_RTP_PACKETIZER_TEST_RTP_HEADER_PARSER_H_