text: /*
   * libjingle
   * Copyright 2010 Google Inc.
   *
   * Redistribution and use in source and binary forms, with or without
   * modification, are permitted provided that the following conditions are met:
   *
   *  1. Redistributions of source code must retain the above copyright notice,
   *     this list of conditions and the following disclaimer.
   *  2. Redistributions in binary form must reproduce the above copyright notice,
   *     this list of conditions and the following disclaimer in the documentation
   *     and/or other materials provided with the distribution.
   *  3. The name of the author may not be used to endorse or promote products
   *     derived from this software without specific prior written permission.
   *
   * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
   * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
   * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
   * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
   * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
   * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
   * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
   * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
   * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
   * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
   */
  
  #ifndef TALK_MEDIA_BASE_RTPDUMP_H_
  #define TALK_MEDIA_BASE_RTPDUMP_H_
  
  #include <string.h>
  
  #include <string>
  #include <vector>
  
  #include "webrtc/base/basictypes.h"
  #include "webrtc/base/bytebuffer.h"
  #include "webrtc/base/stream.h"
  
  namespace cricket {
  
  // We use the RTP dump file format compatible to the format used by rtptools
  // (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark
  // (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the
  // first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header.
  // For each packet, the file contains a 8 byte dump packet header, followed by
  // the actual RTP or RTCP packet.
  
  enum RtpDumpPacketFilter {
    PF_NONE = 0x0,
    PF_RTPHEADER = 0x1,
    PF_RTPPACKET = 0x3,  // includes header
    // PF_RTCPHEADER = 0x4,  // TODO(juberti)
    PF_RTCPPACKET = 0xC,  // includes header
    PF_ALL = 0xF
  };
  
  struct RtpDumpFileHeader {
    RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p);
    void WriteToByteBuffer(rtc::ByteBuffer* buf);
  
    static const char kFirstLine[];
    static const size_t kHeaderLength = 16;
    uint32_t start_sec;   // start of recording, the seconds part.
    uint32_t start_usec;  // start of recording, the microseconds part.
    uint32_t source;      // network source (multicast address).
    uint16_t port;        // UDP port.
    uint16_t padding;     // 2 bytes padding.
  };
  
  struct RtpDumpPacket {
    RtpDumpPacket() {}
  
    RtpDumpPacket(const void* d, size_t s, uint32_t elapsed, bool rtcp)
        : elapsed_time(elapsed), original_data_len((rtcp) ? 0 : s) {
      data.resize(s);
      memcpy(&data[0], d, s);
    }
  
    // In the rtpdump file format, RTCP packets have their data len set to zero,
    // since RTCP has an internal length field.
    bool is_rtcp() const { return original_data_len == 0; }
    bool IsValidRtpPacket() const;
    bool IsValidRtcpPacket() const;
    // Get the payload type, sequence number, timestampe, and SSRC of the RTP
    // packet. Return true and set the output parameter if successful.
    bool GetRtpPayloadType(int* pt) const;
    bool GetRtpSeqNum(int* seq_num) const;
    bool GetRtpTimestamp(uint32_t* ts) const;
    bool GetRtpSsrc(uint32_t* ssrc) const;
    bool GetRtpHeaderLen(size_t* len) const;
    // Get the type of the RTCP packet. Return true and set the output parameter
    // if successful.
    bool GetRtcpType(int* type) const;
  
    static const size_t kHeaderLength = 8;
    uint32_t elapsed_time;      // Milliseconds since the start of recording.
    std::vector<uint8_t> data;  // The actual RTP or RTCP packet.
    size_t original_data_len;  // The original length of the packet; may be
                               // greater than data.size() if only part of the
                               // packet was recorded.
  };
  
  class RtpDumpReader {
   public:
    explicit RtpDumpReader(rtc::StreamInterface* stream)
        : stream_(stream),
          file_header_read_(false),
          first_line_and_file_header_len_(0),
          start_time_ms_(0),
          ssrc_override_(0) {
    }
    virtual ~RtpDumpReader() {}
  
    // Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
    void SetSsrc(uint32_t ssrc);
    virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
  
   protected:
    rtc::StreamResult ReadFileHeader();
    bool RewindToFirstDumpPacket() {
      return stream_->SetPosition(first_line_and_file_header_len_);
    }
  
   private:
    // Check if its matches "#!rtpplay1.0 address/port\n".
    bool CheckFirstLine(const std::string& first_line);
  
    rtc::StreamInterface* stream_;
    bool file_header_read_;
    size_t first_line_and_file_header_len_;
    uint32_t start_time_ms_;
    uint32_t ssrc_override_;
  
    RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
  };
  
  // RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
  // the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
  // RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
  // handle both RTP dump and RTCP dump. We assume that the dump does not mix
  // RTP packets and RTCP packets.
  class RtpDumpLoopReader : public RtpDumpReader {
   public:
    explicit RtpDumpLoopReader(rtc::StreamInterface* stream);
    virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
  
   private:
    // During the first loop, update the statistics, including packet count, frame
    // count, timestamps, and sequence number, of the input stream.
    void UpdateStreamStatistics(const RtpDumpPacket& packet);
  
    // At the end of first loop, calculate elapsed_time_increases_,
    // rtp_seq_num_increase_, and rtp_timestamp_increase_.
    void CalculateIncreases();
  
    // During the second and later loops, update the elapsed time of the dump
    // packet. If the dumped packet is a RTP packet, update its RTP sequence
    // number and timestamp as well.
    void UpdateDumpPacket(RtpDumpPacket* packet);
  
    int loop_count_;
    // How much to increase the elapsed time, RTP sequence number, RTP timestampe
    // for each loop. They are calcualted with the variables below during the
    // first loop.
    uint32_t elapsed_time_increases_;
    int rtp_seq_num_increase_;
    uint32_t rtp_timestamp_increase_;
    // How many RTP packets and how many payload frames in the input stream. RTP
    // packets belong to the same frame have the same RTP timestamp, different
    // dump timestamp, and different RTP sequence number.
    uint32_t packet_count_;
    uint32_t frame_count_;
    // The elapsed time, RTP sequence number, and RTP timestamp of the first and
    // the previous dump packets in the input stream.
    uint32_t first_elapsed_time_;
    int first_rtp_seq_num_;
    uint32_t first_rtp_timestamp_;
    uint32_t prev_elapsed_time_;
    int prev_rtp_seq_num_;
    uint32_t prev_rtp_timestamp_;
  
    RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
  };
  
  class RtpDumpWriter {
   public:
    explicit RtpDumpWriter(rtc::StreamInterface* stream);
  
    // Filter to control what packets we actually record.
    void set_packet_filter(int filter);
    // Write a RTP or RTCP packet. The parameters data points to the packet and
    // data_len is its length.
    rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
      return WritePacket(data, data_len, GetElapsedTime(), false);
    }
    rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
      return WritePacket(data, data_len, GetElapsedTime(), true);
    }
    rtc::StreamResult WritePacket(const RtpDumpPacket& packet) {
      return WritePacket(&packet.data[0], packet.data.size(), packet.elapsed_time,
                         packet.is_rtcp());
    }
    uint32_t GetElapsedTime() const;
  
    bool GetDumpSize(size_t* size) {
      // Note that we use GetPosition(), rather than GetSize(), to avoid flush the
      // stream per write.
      return stream_ && size && stream_->GetPosition(size);
    }
  
   protected:
    rtc::StreamResult WriteFileHeader();
  
   private:
    rtc::StreamResult WritePacket(const void* data,
                                  size_t data_len,
                                  uint32_t elapsed,
                                  bool rtcp);
    size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
    rtc::StreamResult WriteToStream(const void* data, size_t data_len);
  
    rtc::StreamInterface* stream_;
    int packet_filter_;
    bool file_header_written_;
    uint32_t start_time_ms_;  // Time when the record starts.
    // If writing to the stream takes longer than this many ms, log a warning.
    uint32_t warn_slow_writes_delay_;
    RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
  };
  
  }  // namespace cricket
  
  #endif  // TALK_MEDIA_BASE_RTPDUMP_H_
  