modules/audio_coding/test/EncodeDecodeTest.cc - src/webrtc - Git at Google

 /*
  *  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/test/EncodeDecodeTest.h"

 #include <memory>
 #include <sstream>
 #include <stdio.h>
 #include <stdlib.h>

 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/gtest.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, size_t 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 test length to 500 ms (50 blocks of 10 ms each).
   _pcmFile.SetNum10MsBlocksToRead(50);
   // Fast-forward 1 second (100 blocks) since the file starts with silence.
   _pcmFile.FastForward(100);

   // 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, size_t 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(true, acm->RegisterReceiveCodec(recvCodec.pltype,
                                                 CodecInstToSdp(recvCodec)));
     // Forces mono/stereo for Opus.
     if (!strcmp(recvCodec.plname, "opus")) {
       recvCodec.channels = channels;
       EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
                                                 CodecInstToSdp(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;
   bool muted;
   int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
   if (muted) {
     ADD_FAILURE();
     return false;
   }
   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;

   std::unique_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) {
   std::unique_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
	/*
	* 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/test/EncodeDecodeTest.h"

	#include <memory>
	#include <sstream>
	#include <stdio.h>
	#include <stdlib.h>

	#include "webrtc/common_types.h"
	#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
	#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
	#include "webrtc/modules/audio_coding/test/utility.h"
	#include "webrtc/system_wrappers/include/trace.h"
	#include "webrtc/test/gtest.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, size_t 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 test length to 500 ms (50 blocks of 10 ms each).
	_pcmFile.SetNum10MsBlocksToRead(50);
	// Fast-forward 1 second (100 blocks) since the file starts with silence.
	_pcmFile.FastForward(100);

	// 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, size_t 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(true, acm->RegisterReceiveCodec(recvCodec.pltype,
	CodecInstToSdp(recvCodec)));
	// Forces mono/stereo for Opus.
	if (!strcmp(recvCodec.plname, "opus")) {
	recvCodec.channels = channels;
	EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
	CodecInstToSdp(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;
	bool muted;
	int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
	if (muted) {
	ADD_FAILURE();
	return false;
	}
	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;

	std::unique_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) {
	std::unique_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