1/*
2 * libjingle
3 * Copyright 2010, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 *  1. Redistributions of source code must retain the above copyright notice,
9 *     this list of conditions and the following disclaimer.
10 *  2. Redistributions in binary form must reproduce the above copyright notice,
11 *     this list of conditions and the following disclaimer in the documentation
12 *     and/or other materials provided with the distribution.
13 *  3. The name of the author may not be used to endorse or promote products
14 *     derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_SESSION_PHONE_RTPDUMP_H_
29#define TALK_SESSION_PHONE_RTPDUMP_H_
30
31#include <cstring>
32#include <string>
33#include <vector>
34
35#include "talk/base/basictypes.h"
36#include "talk/base/stream.h"
37
38namespace talk_base {
39class ByteBuffer;
40}
41
42namespace cricket {
43
44// We use the RTP dump file format compatible to the format used by rtptools
45// (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark
46// (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the
47// first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header.
48// For each packet, the file contains a 8 byte dump packet header, followed by
49// the actual RTP or RTCP packet.
50
51struct RtpDumpFileHeader {
52  RtpDumpFileHeader(uint32 start_ms, uint32 s, uint16 p);
53  void WriteToByteBuffer(talk_base::ByteBuffer* buf);
54
55  static const std::string kFirstLine;
56  static const size_t kHeaderLength = 16;
57  uint32 start_sec;   // start of recording, the seconds part.
58  uint32 start_usec;  // start of recording, the microseconds part.
59  uint32 source;      // network source (multicast address).
60  uint16 port;        // UDP port.
61  uint16 padding;     // 2 bytes padding.
62};
63
64struct RtpDumpPacket {
65  RtpDumpPacket() {}
66
67  RtpDumpPacket(const void* d, size_t s, uint32 elapsed, bool rtcp)
68      : elapsed_time(elapsed),
69        is_rtcp(rtcp) {
70    data.resize(s);
71    memcpy(&data[0], d, s);
72  }
73
74  bool IsValidRtpPacket() const;
75  // Get the sequence number, timestampe, and SSRC of the RTP packet. Return
76  // true and set the output parameter if successful.
77  bool GetRtpSeqNum(uint16* seq_num) const;
78  bool GetRtpTimestamp(uint32* ts) const;
79  bool GetRtpSsrc(uint32* ssrc) const;
80
81  static const size_t kHeaderLength = 8;
82  uint32 elapsed_time;      // Milliseconds since the start of recording.
83  bool is_rtcp;             // True if the data below is a RTCP packet.
84  std::vector<uint8> data;  // The actual RTP or RTCP packet.
85};
86
87class RtpDumpReader {
88 public:
89  explicit RtpDumpReader(talk_base::StreamInterface* stream)
90      : stream_(stream),
91        file_header_read_(false),
92        first_line_and_file_header_len_(0),
93        start_time_ms_(0) {
94  }
95  virtual ~RtpDumpReader() {}
96
97  virtual talk_base::StreamResult ReadPacket(RtpDumpPacket* packet);
98
99 protected:
100  talk_base::StreamResult ReadFileHeader();
101  bool RewindToFirstDumpPacket() {
102    return stream_->SetPosition(first_line_and_file_header_len_);
103  }
104
105 private:
106  // Check if its matches "#!rtpplay1.0 address/port\n".
107  bool CheckFirstLine(const std::string& first_line);
108
109  talk_base::StreamInterface* stream_;
110  bool file_header_read_;
111  size_t first_line_and_file_header_len_;
112  uint32 start_time_ms_;
113  DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
114};
115
116// RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
117// the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
118// RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
119// handle both RTP dump and RTCP dump. We assume that the dump does not mix
120// RTP packets and RTCP packets.
121class RtpDumpLoopReader : public RtpDumpReader {
122 public:
123  explicit RtpDumpLoopReader(talk_base::StreamInterface* stream);
124  virtual talk_base::StreamResult ReadPacket(RtpDumpPacket* packet);
125
126 private:
127  // During the first loop, update the statistics, including packet count, frame
128  // count, timestamps, and sequence number, of the input stream.
129  void UpdateStreamStatistics(const RtpDumpPacket& packet);
130
131  // At the end of first loop, calculate elapsed_time_increases_,
132  // rtp_seq_num_increase_, and rtp_timestamp_increase_.
133  void CalculateIncreases();
134
135  // During the second and later loops, update the elapsed time of the dump
136  // packet. If the dumped packet is a RTP packet, update its RTP sequence
137  // number and timestamp as well.
138  void UpdateDumpPacket(RtpDumpPacket* packet);
139
140  int loop_count_;
141  // How much to increase the elapsed time, RTP sequence number, RTP timestampe
142  // for each loop. They are calcualted with the variables below during the
143  // first loop.
144  uint32 elapsed_time_increases_;
145  uint16 rtp_seq_num_increase_;
146  uint32 rtp_timestamp_increase_;
147  // How many RTP packets and how many payload frames in the input stream. RTP
148  // packets belong to the same frame have the same RTP timestamp, different
149  // dump timestamp, and different RTP sequence number.
150  uint32 packet_count_;
151  uint32 frame_count_;
152  // The elapsed time, RTP sequence number, and RTP timestamp of the first and
153  // the previous dump packets in the input stream.
154  uint32 first_elapsed_time_;
155  uint16 first_rtp_seq_num_;
156  uint32 first_rtp_timestamp_;
157  uint32 prev_elapsed_time_;
158  uint16 prev_rtp_seq_num_;
159  uint32 prev_rtp_timestamp_;
160
161  DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
162};
163
164class RtpDumpWriter {
165 public:
166  explicit RtpDumpWriter(talk_base::StreamInterface* stream);
167
168  // Write a RTP or RTCP packet. The parameters data points to the packet and
169  // data_len is its length.
170  talk_base::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
171    return WritePacket(data, data_len, GetElapsedTime(), false);
172  }
173  talk_base::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
174    return WritePacket(data, data_len, GetElapsedTime(), true);
175  }
176  talk_base::StreamResult WritePacket(const RtpDumpPacket& packet) {
177    return WritePacket(&packet.data[0], packet.data.size(), packet.elapsed_time,
178                       packet.is_rtcp);
179  }
180  uint32 GetElapsedTime() const;
181
182  bool GetDumpSize(size_t* size) {
183    // Note that we use GetPosition(), rather than GetSize(), to avoid flush the
184    // stream per write.
185    return stream_ && size && stream_->GetPosition(size);
186  }
187
188 protected:
189  talk_base::StreamResult WriteFileHeader();
190
191 private:
192  talk_base::StreamResult WritePacket(const void* data, size_t data_len,
193                                      uint32 elapsed, bool rtcp);
194
195  talk_base::StreamInterface* stream_;
196  bool file_header_written_;
197  uint32 start_time_ms_;  // Time when the record starts.
198  DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
199};
200
201}  // namespace cricket
202
203#endif  // TALK_SESSION_PHONE_RTPDUMP_H_
204