audio_coding: remove "main" directory

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

Index: webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
deleted file mode 100644
index d68e57532e0d92e715f7f60674e248b443deb4db..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ /dev/null
@@ -1,351 +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 "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
-
-#include <sstream>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
-    : _rtpStream(rtpStream),
-      _frequency(frequency),
-      _seqNo(0) {
-}
-
-TestPacketization::~TestPacketization() {
-}
-
-int32_t TestPacketization::SendData(
-    const FrameType /* frameType */, const uint8_t payloadType,
-    const uint32_t timeStamp, const uint8_t* payloadData,
-    const size_t payloadSize,
-    const RTPFragmentationHeader* /* fragmentation */) {
-  _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
-                    _frequency);
-  return 1;
-}
-
-Sender::Sender()
-    : _acm(NULL),
-      _pcmFile(),
-      _audioFrame(),
-      _packetization(NULL) {
-}
-
-void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                   std::string in_file_name, int sample_rate, int channels) {
-  struct CodecInst sendCodec;
-  int noOfCodecs = acm->NumberOfCodecs();
-  int codecNo;
-
-  // Open input file
-  const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
-  _pcmFile.Open(file_name, sample_rate, "rb");
-  if (channels == 2) {
-    _pcmFile.ReadStereo(true);
-  }
-
-  // Set the codec for the current test.
-  if ((testMode == 0) || (testMode == 1)) {
-    // Set the codec id.
-    codecNo = codeId;
-  } else {
-    // Choose codec on command line.
-    printf("List of supported codec.\n");
-    for (int n = 0; n < noOfCodecs; n++) {
-      EXPECT_EQ(0, acm->Codec(n, &sendCodec));
-      printf("%d %s\n", n, sendCodec.plname);
-    }
-    printf("Choose your codec:");
-    ASSERT_GT(scanf("%d", &codecNo), 0);
-  }
-
-  EXPECT_EQ(0, acm->Codec(codecNo, &sendCodec));
-
-  sendCodec.channels = channels;
-
-  EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
-  _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
-  EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
-
-  _acm = acm;
-}
-
-void Sender::Teardown() {
-  _pcmFile.Close();
-  delete _packetization;
-}
-
-bool Sender::Add10MsData() {
-  if (!_pcmFile.EndOfFile()) {
-    EXPECT_GT(_pcmFile.Read10MsData(_audioFrame), 0);
-    int32_t ok = _acm->Add10MsData(_audioFrame);
-    EXPECT_GE(ok, 0);
-    return ok >= 0 ? true : false;
-  }
-  return false;
-}
-
-void Sender::Run() {
-  while (true) {
-    if (!Add10MsData()) {
-      break;
-    }
-  }
-}
-
-Receiver::Receiver()
-    : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
-      _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
-}
-
-void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                     std::string out_file_name, int channels) {
-  struct CodecInst recvCodec = CodecInst();
-  int noOfCodecs;
-  EXPECT_EQ(0, acm->InitializeReceiver());
-
-  noOfCodecs = acm->NumberOfCodecs();
-  for (int i = 0; i < noOfCodecs; i++) {
-    EXPECT_EQ(0, acm->Codec(i, &recvCodec));
-    if (recvCodec.channels == channels)
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
-    // Forces mono/stereo for Opus.
-    if (!strcmp(recvCodec.plname, "opus")) {
-      recvCodec.channels = channels;
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
-    }
-  }
-
-  int playSampFreq;
-  std::string file_name;
-  std::stringstream file_stream;
-  file_stream << webrtc::test::OutputPath() << out_file_name
-      << static_cast<int>(codeId) << ".pcm";
-  file_name = file_stream.str();
-  _rtpStream = rtpStream;
-
-  if (testMode == 1) {
-    playSampFreq = recvCodec.plfreq;
-    _pcmFile.Open(file_name, recvCodec.plfreq, "wb+");
-  } else if (testMode == 0) {
-    playSampFreq = 32000;
-    _pcmFile.Open(file_name, 32000, "wb+");
-  } else {
-    printf("\nValid output frequencies:\n");
-    printf("8000\n16000\n32000\n-1,");
-    printf("which means output frequency equal to received signal frequency");
-    printf("\n\nChoose output sampling frequency: ");
-    ASSERT_GT(scanf("%d", &playSampFreq), 0);
-    file_name = webrtc::test::OutputPath() + out_file_name + ".pcm";
-    _pcmFile.Open(file_name, playSampFreq, "wb+");
-  }
-
-  _realPayloadSizeBytes = 0;
-  _playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
-  _frequency = playSampFreq;
-  _acm = acm;
-  _firstTime = true;
-}
-
-void Receiver::Teardown() {
-  delete[] _playoutBuffer;
-  _pcmFile.Close();
-  if (testMode > 1) {
-    Trace::ReturnTrace();
-  }
-}
-
-bool Receiver::IncomingPacket() {
-  if (!_rtpStream->EndOfFile()) {
-    if (_firstTime) {
-      _firstTime = false;
-      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
-                                               _payloadSizeBytes, &_nextTime);
-      if (_realPayloadSizeBytes == 0) {
-        if (_rtpStream->EndOfFile()) {
-          _firstTime = true;
-          return true;
-        } else {
-          return false;
-        }
-      }
-    }
-
-    EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
-                                      _rtpInfo));
-    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
-                                             _payloadSizeBytes, &_nextTime);
-    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
-      _firstTime = true;
-    }
-  }
-  return true;
-}
-
-bool Receiver::PlayoutData() {
-  AudioFrame audioFrame;
-
-  int32_t ok =_acm->PlayoutData10Ms(_frequency, &audioFrame);
-  EXPECT_EQ(0, ok);
-  if (ok < 0){
-    return false;
-  }
-  if (_playoutLengthSmpls == 0) {
-    return false;
-  }
-  _pcmFile.Write10MsData(audioFrame.data_,
-      audioFrame.samples_per_channel_ * audioFrame.num_channels_);
-  return true;
-}
-
-void Receiver::Run() {
-  uint8_t counter500Ms = 50;
-  uint32_t clock = 0;
-
-  while (counter500Ms > 0) {
-    if (clock == 0 || clock >= _nextTime) {
-      EXPECT_TRUE(IncomingPacket());
-      if (clock == 0) {
-        clock = _nextTime;
-      }
-    }
-    if ((clock % 10) == 0) {
-      if (!PlayoutData()) {
-        clock++;
-        continue;
-      }
-    }
-    if (_rtpStream->EndOfFile()) {
-      counter500Ms--;
-    }
-    clock++;
-  }
-}
-
-EncodeDecodeTest::EncodeDecodeTest() {
-  _testMode = 2;
-  Trace::CreateTrace();
-  Trace::SetTraceFile(
-      (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
-}
-
-EncodeDecodeTest::EncodeDecodeTest(int testMode) {
-  //testMode == 0 for autotest
-  //testMode == 1 for testing all codecs/parameters
-  //testMode > 1 for specific user-input test (as it was used before)
-  _testMode = testMode;
-  if (_testMode != 0) {
-    Trace::CreateTrace();
-    Trace::SetTraceFile(
-        (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
-  }
-}
-
-void EncodeDecodeTest::Perform() {
-  int numCodecs = 1;
-  int codePars[3];  // Frequency, packet size, rate.
-  int numPars[52];  // Number of codec parameters sets (freq, pacsize, rate)
-                    // to test, for a given codec.
-
-  codePars[0] = 0;
-  codePars[1] = 0;
-  codePars[2] = 0;
-
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
-  struct CodecInst sendCodecTmp;
-  numCodecs = acm->NumberOfCodecs();
-
-  if (_testMode != 2) {
-    for (int n = 0; n < numCodecs; n++) {
-      EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
-      if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
-        numPars[n] = 0;
-      } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
-        numPars[n] = 0;
-      } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
-        numPars[n] = 0;
-      } else if (sendCodecTmp.channels == 2) {
-        numPars[n] = 0;
-      } else {
-        numPars[n] = 1;
-      }
-    }
-  } else {
-    numCodecs = 1;
-    numPars[0] = 1;
-  }
-
-  _receiver.testMode = _testMode;
-
-  // Loop over all mono codecs:
-  for (int codeId = 0; codeId < numCodecs; codeId++) {
-    // Only encode using real mono encoders, not telephone-event and cng.
-    for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
-      // Encode all data to file.
-      std::string fileName = EncodeToFile(1, codeId, codePars, _testMode);
-
-      RTPFile rtpFile;
-      rtpFile.Open(fileName.c_str(), "rb");
-
-      _receiver.codeId = codeId;
-
-      rtpFile.ReadHeader();
-      _receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1);
-      _receiver.Run();
-      _receiver.Teardown();
-      rtpFile.Close();
-    }
-  }
-
-  // End tracing.
-  if (_testMode == 1) {
-    Trace::ReturnTrace();
-  }
-}
-
-std::string EncodeDecodeTest::EncodeToFile(int fileType,
-                                           int codeId,
-                                           int* codePars,
-                                           int testMode) {
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
-  RTPFile rtpFile;
-  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
-                                                    "encode_decode_rtp");
-  rtpFile.Open(fileName.c_str(), "wb+");
-  rtpFile.WriteHeader();
-
-  // Store for auto_test and logging.
-  _sender.testMode = testMode;
-  _sender.codeId = codeId;
-
-  _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
-  if (acm->SendCodec()) {
-    _sender.Run();
-  }
-  _sender.Teardown();
-  rtpFile.Close();
-
-  return fileName;
-}
-
-}  // namespace webrtc