1/*
2 *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
13
14#include <stdio.h>
15#include <string>
16
17#include "webrtc/base/constructormagic.h"
18#include "webrtc/common_types.h"
19#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
20#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
21#include "webrtc/system_wrappers/interface/scoped_ptr.h"
22
23namespace webrtc {
24
25class RtpHeaderParser;
26
27namespace test {
28
29class RtpFileSource : public PacketSource {
30 public:
31  // Creates an RtpFileSource reading from |file_name|. If the file cannot be
32  // opened, or has the wrong format, NULL will be returned.
33  static RtpFileSource* Create(const std::string& file_name);
34
35  virtual ~RtpFileSource();
36
37  // Registers an RTP header extension and binds it to |id|.
38  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
39
40  // Returns a pointer to the next packet. Returns NULL if end of file was
41  // reached, or if a the data was corrupt.
42  virtual Packet* NextPacket();
43
44  // Returns true if the end of file has been reached.
45  virtual bool EndOfFile() const;
46
47 private:
48  static const int kFirstLineLength = 40;
49  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
50  static const size_t kPacketHeaderSize = 8;
51
52  RtpFileSource();
53
54  bool OpenFile(const std::string& file_name);
55
56  bool SkipFileHeader();
57
58  FILE* in_file_;
59  int64_t file_end_;
60  scoped_ptr<RtpHeaderParser> parser_;
61
62  DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
63};
64
65}  // namespace test
66}  // namespace webrtc
67#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
68