Move talk/media to webrtc/media

talk/media/base/rtpdump.h

Index: talk/media/base/rtpdump.h
diff --git a/talk/media/base/rtpdump.h b/talk/media/base/rtpdump.h
deleted file mode 100644
index 0f3091a59e029b7d9bc27e8bcbe6d9c9be4c705a..0000000000000000000000000000000000000000
--- a/talk/media/base/rtpdump.h
+++ /dev/null
@@ -1,234 +0,0 @@
-/*
- * 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_