Move talk/media to webrtc/media

talk/media/base/rtpdump.cc

Index: talk/media/base/rtpdump.cc
diff --git a/talk/media/base/rtpdump.cc b/talk/media/base/rtpdump.cc
deleted file mode 100644
index 6861636c41db101d627df3aea467bb78ead7ba57..0000000000000000000000000000000000000000
--- a/talk/media/base/rtpdump.cc
+++ /dev/null
@@ -1,426 +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.
- */
-
-#include "talk/media/base/rtpdump.h"
-
-#include <ctype.h>
-
-#include <string>
-
-#include "talk/media/base/rtputils.h"
-#include "webrtc/base/byteorder.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/timeutils.h"
-
-namespace {
-static const int kRtpSsrcOffset = 8;
-const int  kWarnSlowWritesDelayMs = 50;
-}  // namespace
-
-namespace cricket {
-
-const char RtpDumpFileHeader::kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
-
-RtpDumpFileHeader::RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p)
-    : start_sec(start_ms / 1000),
-      start_usec(start_ms % 1000 * 1000),
-      source(s),
-      port(p),
-      padding(0) {
-}
-
-void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBuffer* buf) {
-  buf->WriteUInt32(start_sec);
-  buf->WriteUInt32(start_usec);
-  buf->WriteUInt32(source);
-  buf->WriteUInt16(port);
-  buf->WriteUInt16(padding);
-}
-
-static const uint32_t kDefaultTimeIncrease = 30;
-
-bool RtpDumpPacket::IsValidRtpPacket() const {
-  return original_data_len >= data.size() &&
-      data.size() >= kMinRtpPacketLen;
-}
-
-bool RtpDumpPacket::IsValidRtcpPacket() const {
-  return original_data_len == 0 &&
-      data.size() >= kMinRtcpPacketLen;
-}
-
-bool RtpDumpPacket::GetRtpPayloadType(int* pt) const {
-  return IsValidRtpPacket() &&
-      cricket::GetRtpPayloadType(&data[0], data.size(), pt);
-}
-
-bool RtpDumpPacket::GetRtpSeqNum(int* seq_num) const {
-  return IsValidRtpPacket() &&
-      cricket::GetRtpSeqNum(&data[0], data.size(), seq_num);
-}
-
-bool RtpDumpPacket::GetRtpTimestamp(uint32_t* ts) const {
-  return IsValidRtpPacket() &&
-      cricket::GetRtpTimestamp(&data[0], data.size(), ts);
-}
-
-bool RtpDumpPacket::GetRtpSsrc(uint32_t* ssrc) const {
-  return IsValidRtpPacket() &&
-      cricket::GetRtpSsrc(&data[0], data.size(), ssrc);
-}
-
-bool RtpDumpPacket::GetRtpHeaderLen(size_t* len) const {
-  return IsValidRtpPacket() &&
-      cricket::GetRtpHeaderLen(&data[0], data.size(), len);
-}
-
-bool RtpDumpPacket::GetRtcpType(int* type) const {
-  return IsValidRtcpPacket() &&
-      cricket::GetRtcpType(&data[0], data.size(), type);
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of RtpDumpReader.
-///////////////////////////////////////////////////////////////////////////
-
-void RtpDumpReader::SetSsrc(uint32_t ssrc) {
-  ssrc_override_ = ssrc;
-}
-
-rtc::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) {
-  if (!packet) return rtc::SR_ERROR;
-
-  rtc::StreamResult res = rtc::SR_SUCCESS;
-  // Read the file header if it has not been read yet.
-  if (!file_header_read_) {
-    res = ReadFileHeader();
-    if (res != rtc::SR_SUCCESS) {
-      return res;
-    }
-    file_header_read_ = true;
-  }
-
-  // Read the RTP dump packet header.
-  char header[RtpDumpPacket::kHeaderLength];
-  res = stream_->ReadAll(header, sizeof(header), NULL, NULL);
-  if (res != rtc::SR_SUCCESS) {
-    return res;
-  }
-  rtc::ByteBuffer buf(header, sizeof(header));
-  uint16_t dump_packet_len;
-  uint16_t data_len;
-  // Read the full length of the rtpdump packet, including the rtpdump header.
-  buf.ReadUInt16(&dump_packet_len);
-  packet->data.resize(dump_packet_len - sizeof(header));
-  // Read the size of the original packet, which may be larger than the size in
-  // the rtpdump file, in the event that only part of the packet (perhaps just
-  // the header) was recorded. Note that this field is set to zero for RTCP
-  // packets, which have their own internal length field.
-  buf.ReadUInt16(&data_len);
-  packet->original_data_len = data_len;
-  // Read the elapsed time for this packet (different than RTP timestamp).
-  buf.ReadUInt32(&packet->elapsed_time);
-
-  // Read the actual RTP or RTCP packet.
-  res = stream_->ReadAll(&packet->data[0], packet->data.size(), NULL, NULL);
-
-  // If the packet is RTP and we have specified a ssrc, replace the RTP ssrc
-  // with the specified ssrc.
-  if (res == rtc::SR_SUCCESS &&
-      packet->IsValidRtpPacket() &&
-      ssrc_override_ != 0) {
-    rtc::SetBE32(&packet->data[kRtpSsrcOffset], ssrc_override_);
-  }
-
-  return res;
-}
-
-rtc::StreamResult RtpDumpReader::ReadFileHeader() {
-  // Read the first line.
-  std::string first_line;
-  rtc::StreamResult res = stream_->ReadLine(&first_line);
-  if (res != rtc::SR_SUCCESS) {
-    return res;
-  }
-  if (!CheckFirstLine(first_line)) {
-    return rtc::SR_ERROR;
-  }
-
-  // Read the 16 byte file header.
-  char header[RtpDumpFileHeader::kHeaderLength];
-  res = stream_->ReadAll(header, sizeof(header), NULL, NULL);
-  if (res == rtc::SR_SUCCESS) {
-    rtc::ByteBuffer buf(header, sizeof(header));
-    uint32_t start_sec;
-    uint32_t start_usec;
-    buf.ReadUInt32(&start_sec);
-    buf.ReadUInt32(&start_usec);
-    start_time_ms_ = start_sec * 1000 + start_usec / 1000;
-    // Increase the length by 1 since first_line does not contain the ending \n.
-    first_line_and_file_header_len_ = first_line.size() + 1 + sizeof(header);
-  }
-  return res;
-}
-
-bool RtpDumpReader::CheckFirstLine(const std::string& first_line) {
-  // The first line is like "#!rtpplay1.0 address/port"
-  bool matched = (0 == first_line.find("#!rtpplay1.0 "));
-
-  // The address could be IP or hostname. We do not check it here. Instead, we
-  // check the port at the end.
-  size_t pos = first_line.find('/');
-  matched &= (pos != std::string::npos && pos < first_line.size() - 1);
-  for (++pos; pos < first_line.size() && matched; ++pos) {
-    matched &= (0 != isdigit(first_line[pos]));
-  }
-
-  return matched;
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of RtpDumpLoopReader.
-///////////////////////////////////////////////////////////////////////////
-RtpDumpLoopReader::RtpDumpLoopReader(rtc::StreamInterface* stream)
-    : RtpDumpReader(stream),
-      loop_count_(0),
-      elapsed_time_increases_(0),
-      rtp_seq_num_increase_(0),
-      rtp_timestamp_increase_(0),
-      packet_count_(0),
-      frame_count_(0),
-      first_elapsed_time_(0),
-      first_rtp_seq_num_(0),
-      first_rtp_timestamp_(0),
-      prev_elapsed_time_(0),
-      prev_rtp_seq_num_(0),
-      prev_rtp_timestamp_(0) {
-}
-
-rtc::StreamResult RtpDumpLoopReader::ReadPacket(RtpDumpPacket* packet) {
-  if (!packet) return rtc::SR_ERROR;
-
-  rtc::StreamResult res = RtpDumpReader::ReadPacket(packet);
-  if (rtc::SR_SUCCESS == res) {
-    if (0 == loop_count_) {
-      // During the first loop, we update the statistics of the input stream.
-      UpdateStreamStatistics(*packet);
-    }
-  } else if (rtc::SR_EOS == res) {
-    if (0 == loop_count_) {
-      // At the end of the first loop, calculate elapsed_time_increases_,
-      // rtp_seq_num_increase_, and rtp_timestamp_increase_, which will be
-      // used during the second and later loops.
-      CalculateIncreases();
-    }
-
-    // Rewind the input stream to the first dump packet and read again.
-    ++loop_count_;
-    if (RewindToFirstDumpPacket()) {
-      res = RtpDumpReader::ReadPacket(packet);
-    }
-  }
-
-  if (rtc::SR_SUCCESS == res && loop_count_ > 0) {
-    // During the second and later loops, we update the elapsed time of the dump
-    // packet. If the dumped packet is a RTP packet, we also update its RTP
-    // sequence number and timestamp.
-    UpdateDumpPacket(packet);
-  }
-
-  return res;
-}
-
-void RtpDumpLoopReader::UpdateStreamStatistics(const RtpDumpPacket& packet) {
-  // Get the RTP sequence number and timestamp of the dump packet.
-  int rtp_seq_num = 0;
-  packet.GetRtpSeqNum(&rtp_seq_num);
-  uint32_t rtp_timestamp = 0;
-  packet.GetRtpTimestamp(&rtp_timestamp);
-
-  // Set the timestamps and sequence number for the first dump packet.
-  if (0 == packet_count_++) {
-    first_elapsed_time_ = packet.elapsed_time;
-    first_rtp_seq_num_ = rtp_seq_num;
-    first_rtp_timestamp_ = rtp_timestamp;
-    // The first packet belongs to a new payload frame.
-    ++frame_count_;
-  } else if (rtp_timestamp != prev_rtp_timestamp_) {
-    // The current and previous packets belong to different payload frames.
-    ++frame_count_;
-  }
-
-  prev_elapsed_time_ = packet.elapsed_time;
-  prev_rtp_timestamp_ = rtp_timestamp;
-  prev_rtp_seq_num_ = rtp_seq_num;
-}
-
-void RtpDumpLoopReader::CalculateIncreases() {
-  // At this time, prev_elapsed_time_, prev_rtp_seq_num_, and
-  // prev_rtp_timestamp_ are values of the last dump packet in the input stream.
-  rtp_seq_num_increase_ = prev_rtp_seq_num_ - first_rtp_seq_num_ + 1;
-  // If we have only one packet or frame, we use the default timestamp
-  // increase. Otherwise, we use the difference between the first and the last
-  // packets or frames.
-  elapsed_time_increases_ = packet_count_ <= 1 ? kDefaultTimeIncrease :
-      (prev_elapsed_time_ - first_elapsed_time_) * packet_count_ /
-      (packet_count_ - 1);
-  rtp_timestamp_increase_ = frame_count_ <= 1 ? kDefaultTimeIncrease :
-      (prev_rtp_timestamp_ - first_rtp_timestamp_) * frame_count_ /
-      (frame_count_ - 1);
-}
-
-void RtpDumpLoopReader::UpdateDumpPacket(RtpDumpPacket* packet) {
-  // Increase the elapsed time of the dump packet.
-  packet->elapsed_time += loop_count_ * elapsed_time_increases_;
-
-  if (packet->IsValidRtpPacket()) {
-    // Get the old RTP sequence number and timestamp.
-    int sequence = 0;
-    packet->GetRtpSeqNum(&sequence);
-    uint32_t timestamp = 0;
-    packet->GetRtpTimestamp(&timestamp);
-    // Increase the RTP sequence number and timestamp.
-    sequence += loop_count_ * rtp_seq_num_increase_;
-    timestamp += loop_count_ * rtp_timestamp_increase_;
-    // Write the updated sequence number and timestamp back to the RTP packet.
-    rtc::ByteBuffer buffer;
-    buffer.WriteUInt16(sequence);
-    buffer.WriteUInt32(timestamp);
-    memcpy(&packet->data[2], buffer.Data(), buffer.Length());
-  }
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of RtpDumpWriter.
-///////////////////////////////////////////////////////////////////////////
-
-RtpDumpWriter::RtpDumpWriter(rtc::StreamInterface* stream)
-    : stream_(stream),
-      packet_filter_(PF_ALL),
-      file_header_written_(false),
-      start_time_ms_(rtc::Time()),
-      warn_slow_writes_delay_(kWarnSlowWritesDelayMs) {
-}
-
-void RtpDumpWriter::set_packet_filter(int filter) {
-  packet_filter_ = filter;
-  LOG(LS_INFO) << "RtpDumpWriter set_packet_filter to " << packet_filter_;
-}
-
-uint32_t RtpDumpWriter::GetElapsedTime() const {
-  return rtc::TimeSince(start_time_ms_);
-}
-
-rtc::StreamResult RtpDumpWriter::WriteFileHeader() {
-  rtc::StreamResult res = WriteToStream(
-      RtpDumpFileHeader::kFirstLine,
-      strlen(RtpDumpFileHeader::kFirstLine));
-  if (res != rtc::SR_SUCCESS) {
-    return res;
-  }
-
-  rtc::ByteBuffer buf;
-  RtpDumpFileHeader file_header(rtc::Time(), 0, 0);
-  file_header.WriteToByteBuffer(&buf);
-  return WriteToStream(buf.Data(), buf.Length());
-}
-
-rtc::StreamResult RtpDumpWriter::WritePacket(const void* data,
-                                             size_t data_len,
-                                             uint32_t elapsed,
-                                             bool rtcp) {
-  if (!stream_ || !data || 0 == data_len) return rtc::SR_ERROR;
-
-  rtc::StreamResult res = rtc::SR_SUCCESS;
-  // Write the file header if it has not been written yet.
-  if (!file_header_written_) {
-    res = WriteFileHeader();
-    if (res != rtc::SR_SUCCESS) {
-      return res;
-    }
-    file_header_written_ = true;
-  }
-
-  // Figure out what to write.
-  size_t write_len = FilterPacket(data, data_len, rtcp);
-  if (write_len == 0) {
-    return rtc::SR_SUCCESS;
-  }
-
-  // Write the dump packet header.
-  rtc::ByteBuffer buf;
-  buf.WriteUInt16(
-      static_cast<uint16_t>(RtpDumpPacket::kHeaderLength + write_len));
-  buf.WriteUInt16(static_cast<uint16_t>(rtcp ? 0 : data_len));
-  buf.WriteUInt32(elapsed);
-  res = WriteToStream(buf.Data(), buf.Length());
-  if (res != rtc::SR_SUCCESS) {
-    return res;
-  }
-
-  // Write the header or full packet as indicated by write_len.
-  return WriteToStream(data, write_len);
-}
-
-size_t RtpDumpWriter::FilterPacket(const void* data, size_t data_len,
-                                   bool rtcp) {
-  size_t filtered_len = 0;
-  if (!rtcp) {
-    if ((packet_filter_ & PF_RTPPACKET) == PF_RTPPACKET) {
-      // RTP header + payload
-      filtered_len = data_len;
-    } else if ((packet_filter_ & PF_RTPHEADER) == PF_RTPHEADER) {
-      // RTP header only
-      size_t header_len;
-      if (GetRtpHeaderLen(data, data_len, &header_len)) {
-        filtered_len = header_len;
-      }
-    }
-  } else {
-    if ((packet_filter_ & PF_RTCPPACKET) == PF_RTCPPACKET) {
-      // RTCP header + payload
-      filtered_len = data_len;
-    }
-  }
-
-  return filtered_len;
-}
-
-rtc::StreamResult RtpDumpWriter::WriteToStream(
-    const void* data, size_t data_len) {
-  uint32_t before = rtc::Time();
-  rtc::StreamResult result =
-      stream_->WriteAll(data, data_len, NULL, NULL);
-  uint32_t delay = rtc::TimeSince(before);
-  if (delay >= warn_slow_writes_delay_) {
-    LOG(LS_WARNING) << "Slow RtpDump: took " << delay << "ms to write "
-                    << data_len << " bytes.";
-  }
-  return result;
-}
-
-}  // namespace cricket