audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/RTPFile.cc

Index: webrtc/modules/audio_coding/main/test/RTPFile.cc
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/main/test/RTPFile.cc
deleted file mode 100644
index 60777178c698dce1298157865942bc9edf3949d9..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- *  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