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_MEDIA_BASE_RTPDUMP_H_
29 #define TALK_MEDIA_BASE_RTPDUMP_H_
30 
31 #include <string.h>
32 
33 #include <string>
34 #include <vector>
35 
36 #include "webrtc/base/basictypes.h"
37 #include "webrtc/base/bytebuffer.h"
38 #include "webrtc/base/stream.h"
39 
40 namespace cricket {
41 
42 // We use the RTP dump file format compatible to the format used by rtptools
43 // (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark
44 // (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the
45 // first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header.
46 // For each packet, the file contains a 8 byte dump packet header, followed by
47 // the actual RTP or RTCP packet.
48 
49 enum RtpDumpPacketFilter {
50   PF_NONE = 0x0,
51   PF_RTPHEADER = 0x1,
52   PF_RTPPACKET = 0x3,  // includes header
53   // PF_RTCPHEADER = 0x4,  // TODO(juberti)
54   PF_RTCPPACKET = 0xC,  // includes header
55   PF_ALL = 0xF
56 };
57 
58 struct RtpDumpFileHeader {
59   RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p);
60   void WriteToByteBuffer(rtc::ByteBuffer* buf);
61 
62   static const char kFirstLine[];
63   static const size_t kHeaderLength = 16;
64   uint32_t start_sec;   // start of recording, the seconds part.
65   uint32_t start_usec;  // start of recording, the microseconds part.
66   uint32_t source;      // network source (multicast address).
67   uint16_t port;        // UDP port.
68   uint16_t padding;     // 2 bytes padding.
69 };
70 
71 struct RtpDumpPacket {
RtpDumpPacketRtpDumpPacket72   RtpDumpPacket() {}
73 
RtpDumpPacketRtpDumpPacket74   RtpDumpPacket(const void* d, size_t s, uint32_t elapsed, bool rtcp)
75       : elapsed_time(elapsed), original_data_len((rtcp) ? 0 : s) {
76     data.resize(s);
77     memcpy(&data[0], d, s);
78   }
79 
80   // In the rtpdump file format, RTCP packets have their data len set to zero,
81   // since RTCP has an internal length field.
is_rtcpRtpDumpPacket82   bool is_rtcp() const { return original_data_len == 0; }
83   bool IsValidRtpPacket() const;
84   bool IsValidRtcpPacket() const;
85   // Get the payload type, sequence number, timestampe, and SSRC of the RTP
86   // packet. Return true and set the output parameter if successful.
87   bool GetRtpPayloadType(int* pt) const;
88   bool GetRtpSeqNum(int* seq_num) const;
89   bool GetRtpTimestamp(uint32_t* ts) const;
90   bool GetRtpSsrc(uint32_t* ssrc) const;
91   bool GetRtpHeaderLen(size_t* len) const;
92   // Get the type of the RTCP packet. Return true and set the output parameter
93   // if successful.
94   bool GetRtcpType(int* type) const;
95 
96   static const size_t kHeaderLength = 8;
97   uint32_t elapsed_time;      // Milliseconds since the start of recording.
98   std::vector<uint8_t> data;  // The actual RTP or RTCP packet.
99   size_t original_data_len;  // The original length of the packet; may be
100                              // greater than data.size() if only part of the
101                              // packet was recorded.
102 };
103 
104 class RtpDumpReader {
105  public:
RtpDumpReader(rtc::StreamInterface * stream)106   explicit RtpDumpReader(rtc::StreamInterface* stream)
107       : stream_(stream),
108         file_header_read_(false),
109         first_line_and_file_header_len_(0),
110         start_time_ms_(0),
111         ssrc_override_(0) {
112   }
~RtpDumpReader()113   virtual ~RtpDumpReader() {}
114 
115   // Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
116   void SetSsrc(uint32_t ssrc);
117   virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
118 
119  protected:
120   rtc::StreamResult ReadFileHeader();
RewindToFirstDumpPacket()121   bool RewindToFirstDumpPacket() {
122     return stream_->SetPosition(first_line_and_file_header_len_);
123   }
124 
125  private:
126   // Check if its matches "#!rtpplay1.0 address/port\n".
127   bool CheckFirstLine(const std::string& first_line);
128 
129   rtc::StreamInterface* stream_;
130   bool file_header_read_;
131   size_t first_line_and_file_header_len_;
132   uint32_t start_time_ms_;
133   uint32_t ssrc_override_;
134 
135   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
136 };
137 
138 // RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
139 // the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
140 // RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
141 // handle both RTP dump and RTCP dump. We assume that the dump does not mix
142 // RTP packets and RTCP packets.
143 class RtpDumpLoopReader : public RtpDumpReader {
144  public:
145   explicit RtpDumpLoopReader(rtc::StreamInterface* stream);
146   virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
147 
148  private:
149   // During the first loop, update the statistics, including packet count, frame
150   // count, timestamps, and sequence number, of the input stream.
151   void UpdateStreamStatistics(const RtpDumpPacket& packet);
152 
153   // At the end of first loop, calculate elapsed_time_increases_,
154   // rtp_seq_num_increase_, and rtp_timestamp_increase_.
155   void CalculateIncreases();
156 
157   // During the second and later loops, update the elapsed time of the dump
158   // packet. If the dumped packet is a RTP packet, update its RTP sequence
159   // number and timestamp as well.
160   void UpdateDumpPacket(RtpDumpPacket* packet);
161 
162   int loop_count_;
163   // How much to increase the elapsed time, RTP sequence number, RTP timestampe
164   // for each loop. They are calcualted with the variables below during the
165   // first loop.
166   uint32_t elapsed_time_increases_;
167   int rtp_seq_num_increase_;
168   uint32_t rtp_timestamp_increase_;
169   // How many RTP packets and how many payload frames in the input stream. RTP
170   // packets belong to the same frame have the same RTP timestamp, different
171   // dump timestamp, and different RTP sequence number.
172   uint32_t packet_count_;
173   uint32_t frame_count_;
174   // The elapsed time, RTP sequence number, and RTP timestamp of the first and
175   // the previous dump packets in the input stream.
176   uint32_t first_elapsed_time_;
177   int first_rtp_seq_num_;
178   uint32_t first_rtp_timestamp_;
179   uint32_t prev_elapsed_time_;
180   int prev_rtp_seq_num_;
181   uint32_t prev_rtp_timestamp_;
182 
183   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
184 };
185 
186 class RtpDumpWriter {
187  public:
188   explicit RtpDumpWriter(rtc::StreamInterface* stream);
189 
190   // Filter to control what packets we actually record.
191   void set_packet_filter(int filter);
192   // Write a RTP or RTCP packet. The parameters data points to the packet and
193   // data_len is its length.
WriteRtpPacket(const void * data,size_t data_len)194   rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
195     return WritePacket(data, data_len, GetElapsedTime(), false);
196   }
WriteRtcpPacket(const void * data,size_t data_len)197   rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
198     return WritePacket(data, data_len, GetElapsedTime(), true);
199   }
WritePacket(const RtpDumpPacket & packet)200   rtc::StreamResult WritePacket(const RtpDumpPacket& packet) {
201     return WritePacket(&packet.data[0], packet.data.size(), packet.elapsed_time,
202                        packet.is_rtcp());
203   }
204   uint32_t GetElapsedTime() const;
205 
GetDumpSize(size_t * size)206   bool GetDumpSize(size_t* size) {
207     // Note that we use GetPosition(), rather than GetSize(), to avoid flush the
208     // stream per write.
209     return stream_ && size && stream_->GetPosition(size);
210   }
211 
212  protected:
213   rtc::StreamResult WriteFileHeader();
214 
215  private:
216   rtc::StreamResult WritePacket(const void* data,
217                                 size_t data_len,
218                                 uint32_t elapsed,
219                                 bool rtcp);
220   size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
221   rtc::StreamResult WriteToStream(const void* data, size_t data_len);
222 
223   rtc::StreamInterface* stream_;
224   int packet_filter_;
225   bool file_header_written_;
226   uint32_t start_time_ms_;  // Time when the record starts.
227   // If writing to the stream takes longer than this many ms, log a warning.
228   uint32_t warn_slow_writes_delay_;
229   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
230 };
231 
232 }  // namespace cricket
233 
234 #endif  // TALK_MEDIA_BASE_RTPDUMP_H_
235