1 files changed, 227 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/RTPFile.cc b/webrtc/modules/audio_coding/test/RTPFile.cc
new file mode 100644
index 0000000000..60777178c6
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/RTPFile.cc
@@ -0,0 +1,227 @@
+/*
+ *  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 "engine_configurations.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+// TODO(tlegrand): Consider removing usage of gtest.
+#include "testing/gtest/include/gtest/gtest.h"
+
+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