DXR is a code search and navigation tool aimed at making sense of large projects. It supports full-text and regex searches as well as structural queries.

Header

Mercurial (c68fe15a81fc)

VCS Links

Line Code
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227
/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "RTPFile.h"

#include <stdlib.h>
#include <limits>

#ifdef WIN32
#   include <Winsock2.h>
#else
#   include <arpa/inet.h>
#endif

#include "audio_coding_module.h"
#include "system_wrappers/include/rw_lock_wrapper.h"
// TODO(tlegrand): Consider removing usage of gtest.
#include "test/gtest.h"
#include "typedefs.h"  // NOLINT(build/include)

namespace webrtc {

void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
                               const uint8_t* rtpHeader) {
  rtpInfo->header.payloadType = rtpHeader[1];
  rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
      rtpHeader[3];
  rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
      (static_cast<uint32_t>(rtpHeader[5]) << 16) |
      (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
  rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
      (static_cast<uint32_t>(rtpHeader[9]) << 16) |
      (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11];
}

void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
                              int16_t seqNo, uint32_t timeStamp,
                              uint32_t ssrc) {
  rtpHeader[0] = 0x80;
  rtpHeader[1] = payloadType;
  rtpHeader[2] = (seqNo >> 8) & 0xFF;
  rtpHeader[3] = seqNo & 0xFF;
  rtpHeader[4] = timeStamp >> 24;
  rtpHeader[5] = (timeStamp >> 16) & 0xFF;
  rtpHeader[6] = (timeStamp >> 8) & 0xFF;
  rtpHeader[7] = timeStamp & 0xFF;
  rtpHeader[8] = ssrc >> 24;
  rtpHeader[9] = (ssrc >> 16) & 0xFF;
  rtpHeader[10] = (ssrc >> 8) & 0xFF;
  rtpHeader[11] = ssrc & 0xFF;
}

RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
                     const uint8_t* payloadData, size_t payloadSize,
                     uint32_t frequency)
    : payloadType(payloadType),
      timeStamp(timeStamp),
      seqNo(seqNo),
      payloadSize(payloadSize),
      frequency(frequency) {
  if (payloadSize > 0) {
    this->payloadData = new uint8_t[payloadSize];
    memcpy(this->payloadData, payloadData, payloadSize);
  }
}

RTPPacket::~RTPPacket() {
  delete[] payloadData;
}

RTPBuffer::RTPBuffer() {
  _queueRWLock = RWLockWrapper::CreateRWLock();
}

RTPBuffer::~RTPBuffer() {
  delete _queueRWLock;
}

void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
                      const int16_t seqNo, const uint8_t* payloadData,
                      const size_t payloadSize, uint32_t frequency) {
  RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
                                    payloadSize, frequency);
  _queueRWLock->AcquireLockExclusive();
  _rtpQueue.push(packet);
  _queueRWLock->ReleaseLockExclusive();
}

size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
                       size_t payloadSize, uint32_t* offset) {
  _queueRWLock->AcquireLockShared();
  RTPPacket *packet = _rtpQueue.front();
  _rtpQueue.pop();
  _queueRWLock->ReleaseLockShared();
  rtpInfo->header.markerBit = 1;
  rtpInfo->header.payloadType = packet->payloadType;
  rtpInfo->header.sequenceNumber = packet->seqNo;
  rtpInfo->header.ssrc = 0;
  rtpInfo->header.timestamp = packet->timeStamp;
  if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
    memcpy(payloadData, packet->payloadData, packet->payloadSize);
  } else {
    return 0;
  }
  *offset = (packet->timeStamp / (packet->frequency / 1000));

  return packet->payloadSize;
}

bool RTPBuffer::EndOfFile() const {
  _queueRWLock->AcquireLockShared();
  bool eof = _rtpQueue.empty();
  _queueRWLock->ReleaseLockShared();
  return eof;
}

void RTPFile::Open(const char *filename, const char *mode) {
  if ((_rtpFile = fopen(filename, mode)) == NULL) {
    printf("Cannot write file %s.\n", filename);
    ADD_FAILURE() << "Unable to write file";
    exit(1);
  }
}

void RTPFile::Close() {
  if (_rtpFile != NULL) {
    fclose(_rtpFile);
    _rtpFile = NULL;
  }
}

void RTPFile::WriteHeader() {
  // Write data in a format that NetEQ and RTP Play can parse
  fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
  uint32_t dummy_variable = 0;
  // should be converted to network endian format, but does not matter when 0
  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
  fflush(_rtpFile);
}

void RTPFile::ReadHeader() {
  uint32_t start_sec, start_usec, source;
  uint16_t port, padding;
  char fileHeader[40];
  EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
  EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
  start_sec = ntohl(start_sec);
  EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
  start_usec = ntohl(start_usec);
  EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
  source = ntohl(source);
  EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
  port = ntohs(port);
  EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
  padding = ntohs(padding);
}

void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
                    const int16_t seqNo, const uint8_t* payloadData,
                    const size_t payloadSize, uint32_t frequency) {
  /* write RTP packet to file */
  uint8_t rtpHeader[12];
  MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
  ASSERT_LE(12 + payloadSize + 8, std::numeric_limits<u_short>::max());
  uint16_t lengthBytes = htons(static_cast<u_short>(12 + payloadSize + 8));
  uint16_t plen = htons(static_cast<u_short>(12 + payloadSize));
  uint32_t offsetMs;

  offsetMs = (timeStamp / (frequency / 1000));
  offsetMs = htonl(offsetMs);
  EXPECT_EQ(1u, fwrite(&lengthBytes, 2, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&plen, 2, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&offsetMs, 4, 1, _rtpFile));
  EXPECT_EQ(1u, fwrite(&rtpHeader, 12, 1, _rtpFile));
  EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
}

size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
                     size_t payloadSize, uint32_t* offset) {
  uint16_t lengthBytes;
  uint16_t plen;
  uint8_t rtpHeader[12];
  size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
  /* Check if we have reached end of file. */
  if ((read_len == 0) && feof(_rtpFile)) {
    _rtpEOF = true;
    return 0;
  }
  EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
  EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
  lengthBytes = ntohs(lengthBytes);
  plen = ntohs(plen);
  *offset = ntohl(*offset);
  EXPECT_GT(plen, 11);

  EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
  ParseRTPHeader(rtpInfo, rtpHeader);
  rtpInfo->type.Audio.isCNG = false;
  rtpInfo->type.Audio.channel = 1;
  EXPECT_EQ(lengthBytes, plen + 8);

  if (plen == 0) {
    return 0;
  }
  if (lengthBytes < 20) {
    return 0;
  }
  if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
    return 0;
  }
  lengthBytes -= 20;
  EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
  return lengthBytes;
}

}  // namespace webrtc