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      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 {
     72   RtpDumpPacket() {}
     73 
     74   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.
     82   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:
    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   }
    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();
    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.
    194   rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
    195     return WritePacket(data, data_len, GetElapsedTime(), false);
    196   }
    197   rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
    198     return WritePacket(data, data_len, GetElapsedTime(), true);
    199   }
    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 
    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