Changes to solve warnings on Mac, issue #178.
Review URL: http://webrtc-codereview.appspot.com/320005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1216 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/audio_coding/main/test/APITest.cpp b/src/modules/audio_coding/main/test/APITest.cpp
index e6d2f99..78ab01b 100644
--- a/src/modules/audio_coding/main/test/APITest.cpp
+++ b/src/modules/audio_coding/main/test/APITest.cpp
@@ -25,6 +25,8 @@
#include "trace.h"
#include "utility.h"
+namespace webrtc {
+
#define TEST_DURATION_SEC 600
#define NUMBER_OF_SENDER_TESTS 6
@@ -32,7 +34,6 @@
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define CHECK_THREAD_NULLITY(myThread, S) if(myThread != NULL){unsigned int i; (myThread)->Start(i);}else{throw S; exit(1);}
-using namespace webrtc;
void
APITest::Wait(WebRtc_UWord32 waitLengthMs)
@@ -1545,3 +1546,6 @@
_acmB->RegisterIncomingMessagesCallback(NULL);
}
}
+
+} // namespace webrtc
+
diff --git a/src/modules/audio_coding/main/test/APITest.h b/src/modules/audio_coding/main/test/APITest.h
index 80f4653..db0a87c 100644
--- a/src/modules/audio_coding/main/test/APITest.h
+++ b/src/modules/audio_coding/main/test/APITest.h
@@ -17,6 +17,8 @@
#include "event_wrapper.h"
#include "utility.h"
+namespace webrtc {
+
enum APITESTAction {TEST_CHANGE_CODEC_ONLY = 0, DTX_TEST = 1};
class APITest : public ACMTest
@@ -170,5 +172,6 @@
int _testNumB;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/Channel.cpp b/src/modules/audio_coding/main/test/Channel.cpp
index 3849367..363b106 100644
--- a/src/modules/audio_coding/main/test/Channel.cpp
+++ b/src/modules/audio_coding/main/test/Channel.cpp
@@ -17,7 +17,7 @@
#include "typedefs.h"
#include "common_types.h"
-using namespace webrtc;
+namespace webrtc {
WebRtc_Word32
Channel::SendData(
@@ -479,3 +479,5 @@
_channelCritSect->Leave();
return rate;
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/Channel.h b/src/modules/audio_coding/main/test/Channel.h
index 0846788..375bec7 100644
--- a/src/modules/audio_coding/main/test/Channel.h
+++ b/src/modules/audio_coding/main/test/Channel.h
@@ -17,6 +17,7 @@
#include "critical_section_wrapper.h"
#include "rw_lock_wrapper.h"
+namespace webrtc {
#define MAX_NUM_PAYLOADS 50
#define MAX_NUM_FRAMESIZES 6
@@ -43,8 +44,6 @@
ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};
-using namespace webrtc;
-
class Channel: public AudioPacketizationCallback
{
public:
@@ -96,7 +95,7 @@
private:
void CalcStatistics(
- WebRtcRTPHeader& rtpInfo,
+ WebRtcRTPHeader& rtpInfo,
WebRtc_UWord16 payloadSize);
AudioCodingModule* _receiverACM;
@@ -121,5 +120,6 @@
WebRtc_UWord64 _totalBytes;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp b/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp
index d46fa87..7670e2a 100644
--- a/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp
+++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.cpp
@@ -10,296 +10,401 @@
#include "EncodeDecodeTest.h"
+#include <stdio.h>
#include <stdlib.h>
#include <string.h>
+#include "audio_coding_module.h"
#include "common_types.h"
#include "gtest/gtest.h"
#include "trace.h"
#include "utility.h"
-Receiver::Receiver()
-:
-_playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
-_payloadSizeBytes(MAX_INCOMING_PAYLOAD)
-{
+namespace webrtc {
+
+TestPacketization::TestPacketization(RTPStream *rtpStream,
+ WebRtc_UWord16 frequency)
+ : _rtpStream(rtpStream),
+ _frequency(frequency),
+ _seqNo(0) {
}
-void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
-{
- struct CodecInst recvCodec;
- int noOfCodecs;
- acm->InitializeReceiver();
+TestPacketization::~TestPacketization() { }
- noOfCodecs = acm->NumberOfCodecs();
- for (int i=0; i < noOfCodecs; i++)
- {
- acm->Codec((WebRtc_UWord8)i, recvCodec);
- if (acm->RegisterReceiveCodec(recvCodec) != 0)
- {
- printf("Unable to register codec: for run: codecId: %d\n", codeId);
- exit(1);
- }
- }
-
- char filename[128];
- _rtpStream = rtpStream;
- int playSampFreq;
+WebRtc_Word32 TestPacketization::SendData(
+ const FrameType /* frameType */,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* /* fragmentation */) {
+ _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
+ _frequency);
+ return 1;
+}
- if (testMode == 1)
- {
- playSampFreq=recvCodec.plfreq;
- //output file for current run
- sprintf(filename,"./src/modules/audio_coding/main/test/out%dFile.pcm",codeId);
- _pcmFile.Open(filename, recvCodec.plfreq, "wb+");
+Sender::Sender()
+ : _acm(NULL),
+ _pcmFile(),
+ _audioFrame(),
+ _payloadSize(0),
+ _timeStamp(0),
+ _packetization(NULL) {
+}
+
+void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
+ acm->InitializeSender();
+ struct CodecInst sendCodec;
+ int noOfCodecs = acm->NumberOfCodecs();
+ int codecNo;
+
+ if (testMode == 1) {
+ // Set the codec, input file, and parameters for the current test.
+ codecNo = codeId;
+ // Use same input file for now.
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ } else if (testMode == 0) {
+ // Set the codec, input file, and parameters for the current test.
+ codecNo = codeId;
+ acm->Codec(codecNo, sendCodec);
+ // Use same input file for now.
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ } else {
+ printf("List of supported codec.\n");
+ for (int n = 0; n < noOfCodecs; n++) {
+ acm->Codec(n, sendCodec);
+ printf("%d %s\n", n, sendCodec.plname);
}
- else if (testMode == 0)
- {
- playSampFreq=32000;
- //output file for current run
- sprintf(filename,"./src/modules/audio_coding/main/test/encodeDecode_out%d.pcm",codeId);
- _pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
- }
- else
- {
- printf("\nValid output frequencies:\n");
- printf("8000\n16000\n32000\n-1, which means output freq equal to received signal freq");
- printf("\n\nChoose output sampling frequency: ");
- ASSERT_GT(scanf("%d", &playSampFreq), 0);
- char fileName[] = "./src/modules/audio_coding/main/test/outFile.pcm";
- _pcmFile.Open(fileName, 32000, "wb+");
- }
-
- _realPayloadSizeBytes = 0;
- _playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
- _frequency = playSampFreq;
+ printf("Choose your codec:");
+ ASSERT_GT(scanf("%d", &codecNo), 0);
+ char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
+ _pcmFile.Open(fileName, 32000, "rb");
+ }
+
+ acm->Codec(codecNo, sendCodec);
+ acm->RegisterSendCodec(sendCodec);
+ _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
+ if (acm->RegisterTransportCallback(_packetization) < 0) {
+ printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
+ codeId);
+ }
+
_acm = acm;
- _firstTime = true;
+ }
+
+void Sender::Teardown() {
+ _pcmFile.Close();
+ delete _packetization;
}
-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 && _rtpStream->EndOfFile())
- {
- _firstTime = true;
- return true;
- }
- }
-
- WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
- if (ok != 0)
- {
- printf("Error when inserting packet to ACM, for run: codecId: %d\n", codeId);
- exit(1);
- }
- _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, _payloadSizeBytes, &_nextTime);
- if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile())
- {
- _firstTime = true;
- }
+bool Sender::Add10MsData() {
+ if (!_pcmFile.EndOfFile()) {
+ _pcmFile.Read10MsData(_audioFrame);
+ WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
+ if (ok != 0) {
+ printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
+ exit(1);
}
return true;
+ }
+ return false;
}
-bool Receiver::PlayoutData()
-{
- AudioFrame audioFrame;
+bool Sender::Process() {
+ WebRtc_Word32 ok = _acm->Process();
+ if (ok < 0) {
+ printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ return true;
+}
- if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0)
- {
- printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n", codeId);
- exit(1);
+void Sender::Run() {
+ while (true) {
+ if (!Add10MsData()) {
+ break;
}
- if (_playoutLengthSmpls == 0)
- {
+ if (!Process()) { // This could be done in a processing thread
+ break;
+ }
+ }
+}
+
+Receiver::Receiver()
+ : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
+ _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
+}
+
+void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
+ struct CodecInst recvCodec;
+ int noOfCodecs;
+ acm->InitializeReceiver();
+
+ noOfCodecs = acm->NumberOfCodecs();
+ for (int i = 0; i < noOfCodecs; i++) {
+ acm->Codec((WebRtc_UWord8) i, recvCodec);
+ if (acm->RegisterReceiveCodec(recvCodec) != 0) {
+ printf("Unable to register codec: for run: codecId: %d\n", codeId);
+ exit(1);
+ }
+ }
+
+ char filename[128];
+ _rtpStream = rtpStream;
+ int playSampFreq;
+
+ if (testMode == 1) {
+ playSampFreq=recvCodec.plfreq;
+ //output file for current run
+ sprintf(filename,"./src/modules/audio_coding/main/test/out%dFile.pcm",
+ codeId);
+ _pcmFile.Open(filename, recvCodec.plfreq, "wb+");
+ } else if (testMode == 0) {
+ playSampFreq=32000;
+ //output file for current run
+ sprintf(filename,
+ "./src/modules/audio_coding/main/test/encodeDecode_out%d.pcm",
+ codeId);
+ _pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
+ } else {
+ printf("\nValid output frequencies:\n");
+ printf("8000\n16000\n32000\n-1,");
+ printf("which means output freq equal to received signal freq");
+ printf("\n\nChoose output sampling frequency: ");
+ ASSERT_GT(scanf("%d", &playSampFreq), 0);
+ char fileName[] = "./src/modules/audio_coding/main/test/outFile.pcm";
+ _pcmFile.Open(fileName, 32000, "wb+");
+ }
+
+ _realPayloadSizeBytes = 0;
+ _playoutBuffer = new WebRtc_Word16[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) {
+ printf("Error in reading incoming payload.\n");
return false;
- }
- _pcmFile.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
- return true;
-}
-
-void Receiver::Run()
-{
- WebRtc_UWord8 counter500Ms = 50;
-
- WebRtc_UWord32 clock = 0;
-
- while (counter500Ms > 0)
- {
- if (clock == 0 || clock >= _nextTime)
- {
- 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("acm_encdec_test.txt");
-}
-
-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("acm_encdec_test.txt");
+ }
+ if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+ _firstTime = true;
+ return true;
+ }
}
-}
-void EncodeDecodeTest::Perform()
-{
- if(_testMode == 0)
- {
- printf("Running Encode/Decode Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- EncodeDecodeTest ----------");
+ WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
+ _realPayloadSizeBytes, _rtpInfo);
+ if (ok != 0) {
+ printf("Error when inserting packet to ACM, for run: codecId: %d\n",
+ codeId);
+ exit(1);
+ }
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+ _payloadSizeBytes, &_nextTime);
+ if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+ _firstTime = true;
}
-
- int numCodecs = 1;
- int codePars[3]; //freq, pacsize, rate
- int playoutFreq[3]; //8, 16, 32k
-
- int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test, for a given codec
-
- codePars[0]=0;
- codePars[1]=0;
- codePars[2]=0;
-
- if (_testMode == 1)
- {
- AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
- struct CodecInst sendCodecTmp;
- numCodecs = acmTmp->NumberOfCodecs();
- printf("List of supported codec.\n");
- for(int n = 0; n < numCodecs; n++)
- {
- acmTmp->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 {
- numPars[n] = 1;
- printf("%d %s\n", n, sendCodecTmp.plname);
- }
- }
- AudioCodingModule::Destroy(acmTmp);
- playoutFreq[1]=16000;
- }
- else if (_testMode == 0)
- {
- AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
- numCodecs = acmTmp->NumberOfCodecs();
- AudioCodingModule::Destroy(acmTmp);
- struct CodecInst dummyCodec;
-
- //chose range of testing for codecs/parameters
- for(int i = 0 ; i < numCodecs ; i++)
- {
- numPars[i] = 1;
- acmTmp->Codec(i, dummyCodec);
- if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0)
- {
- numPars[i] = 0;
- } else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
- numPars[i] = 0;
- } else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
- numPars[i] = 0;
- }
- }
- playoutFreq[1] = 16000;
- }
- else
- {
- numCodecs = 1;
- numPars[0] = 1;
- playoutFreq[1]=16000;
- }
-
- _receiver.testMode = _testMode;
-
- //loop over all codecs:
- for(int codeId=0;codeId<numCodecs;codeId++)
- {
- //only encode using real encoders, not telephone-event anc cn
- for(int loopPars=1;loopPars<=numPars[codeId];loopPars++)
- {
- if (_testMode == 1)
- {
- printf("\n");
- printf("***FOR RUN: codeId: %d\n",codeId);
- printf("\n");
- }
- else if (_testMode == 0)
- {
- printf(".");
- }
-
- EncodeToFileTest::Perform(1, codeId, codePars, _testMode);
-
- AudioCodingModule *acm = AudioCodingModule::Create(10);
- RTPFile rtpFile;
- char fileName[] = "outFile.rtp";
- rtpFile.Open(fileName, "rb");
-
- _receiver.codeId = codeId;
-
- rtpFile.ReadHeader();
- _receiver.Setup(acm, &rtpFile);
- _receiver.Run();
- _receiver.Teardown();
- rtpFile.Close();
- AudioCodingModule::Destroy(acm);
-
- if (_testMode == 1)
- {
- printf("***COMPLETED RUN FOR: codecID: %d ***\n",
- codeId);
- }
- }
- }
- if (_testMode == 0)
- {
- printf("Done!\n");
- }
- if (_testMode == 1) Trace::ReturnTrace();
+ }
+ return true;
}
+bool Receiver::PlayoutData() {
+ AudioFrame audioFrame;
+
+ if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
+ printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
+ codeId);
+ exit(1);
+ }
+ if (_playoutLengthSmpls == 0) {
+ return false;
+ }
+ _pcmFile.Write10MsData(audioFrame._payloadData,
+ audioFrame._payloadDataLengthInSamples);
+ return true;
+}
+
+void Receiver::Run() {
+ WebRtc_UWord8 counter500Ms = 50;
+ WebRtc_UWord32 clock = 0;
+
+ while (counter500Ms > 0) {
+ if (clock == 0 || clock >= _nextTime) {
+ 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("acm_encdec_test.txt");
+}
+
+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("acm_encdec_test.txt");
+ }
+}
+
+void EncodeDecodeTest::Perform() {
+ if (_testMode == 0) {
+ printf("Running Encode/Decode Test");
+ WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ "---------- EncodeDecodeTest ----------");
+ }
+
+ int numCodecs = 1;
+ int codePars[3]; //freq, pacsize, rate
+ int playoutFreq[3]; //8, 16, 32k
+ int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
+ //for a given codec
+
+ codePars[0] = 0;
+ codePars[1] = 0;
+ codePars[2] = 0;
+
+ if (_testMode == 1) {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ struct CodecInst sendCodecTmp;
+ numCodecs = acmTmp->NumberOfCodecs();
+ printf("List of supported codec.\n");
+ for(int n = 0; n < numCodecs; n++) {
+ acmTmp->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 {
+ numPars[n] = 1;
+ printf("%d %s\n", n, sendCodecTmp.plname);
+ }
+ }
+ AudioCodingModule::Destroy(acmTmp);
+ playoutFreq[1] = 16000;
+ } else if (_testMode == 0) {
+ AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
+ numCodecs = acmTmp->NumberOfCodecs();
+ AudioCodingModule::Destroy(acmTmp);
+ struct CodecInst dummyCodec;
+
+ //chose range of testing for codecs/parameters
+ for(int i = 0 ; i < numCodecs ; i++) {
+ numPars[i] = 1;
+ acmTmp->Codec(i, dummyCodec);
+ if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0) {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
+ numPars[i] = 0;
+ } else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
+ numPars[i] = 0;
+ }
+ }
+ playoutFreq[1] = 16000;
+ } else {
+ numCodecs = 1;
+ numPars[0] = 1;
+ playoutFreq[1]=16000;
+ }
+
+ _receiver.testMode = _testMode;
+
+ //loop over all codecs:
+ for (int codeId = 0; codeId < numCodecs; codeId++) {
+ //only encode using real encoders, not telephone-event anc cn
+ for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
+ if (_testMode == 1) {
+ printf("\n");
+ printf("***FOR RUN: codeId: %d\n", codeId);
+ printf("\n");
+ } else if (_testMode == 0) {
+ printf(".");
+ }
+
+ EncodeToFile(1, codeId, codePars, _testMode);
+
+ AudioCodingModule *acm = AudioCodingModule::Create(10);
+ RTPFile rtpFile;
+ char fileName[] = "outFile.rtp";
+ rtpFile.Open(fileName, "rb");
+
+ _receiver.codeId = codeId;
+
+ rtpFile.ReadHeader();
+ _receiver.Setup(acm, &rtpFile);
+ _receiver.Run();
+ _receiver.Teardown();
+ rtpFile.Close();
+ AudioCodingModule::Destroy(acm);
+
+ if (_testMode == 1) {
+ printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId);
+ }
+ }
+ }
+ if (_testMode == 0) {
+ printf("Done!\n");
+ }
+ if (_testMode == 1)
+ Trace::ReturnTrace();
+}
+
+void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
+ int testMode) {
+ AudioCodingModule *acm = AudioCodingModule::Create(0);
+ RTPFile rtpFile;
+ char fileName[] = "outFile.rtp";
+ rtpFile.Open(fileName, "wb+");
+ rtpFile.WriteHeader();
+
+ //for auto_test and logging
+ _sender.testMode = testMode;
+ _sender.codeId = codeId;
+
+ _sender.Setup(acm, &rtpFile);
+ struct CodecInst sendCodecInst;
+ if (acm->SendCodec(sendCodecInst) >= 0) {
+ _sender.Run();
+ }
+ _sender.Teardown();
+ rtpFile.Close();
+ AudioCodingModule::Destroy(acm);
+}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.h b/src/modules/audio_coding/main/test/EncodeDecodeTest.h
index 01172f3..a730fea 100644
--- a/src/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -8,57 +8,110 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef ENCODEDECODETEST_H
-#define ENCODEDECODETEST_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-#include "EncodeToFileTest.h"
+#include <stdio.h>
+
+#include "ACMTest.h"
+#include "audio_coding_module.h"
+#include "RTPFile.h"
+#include "PCMFile.h"
+#include "typedefs.h"
+
+namespace webrtc {
#define MAX_INCOMING_PAYLOAD 8096
-#include "audio_coding_module.h"
-class Receiver
-{
-public:
- Receiver();
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
- void Teardown();
- void Run();
- bool IncomingPacket();
- bool PlayoutData();
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization: public AudioPacketizationCallback {
+ public:
+ TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
+ ~TestPacketization();
+ virtual WebRtc_Word32 SendData(const FrameType frameType,
+ const WebRtc_UWord8 payloadType,
+ const WebRtc_UWord32 timeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
- //for auto_test and logging
- WebRtc_UWord8 codeId;
- WebRtc_UWord8 testMode;
-
-private:
- AudioCodingModule* _acm;
- bool _rtpEOF;
- RTPStream* _rtpStream;
- PCMFile _pcmFile;
- WebRtc_Word16* _playoutBuffer;
- WebRtc_UWord16 _playoutLengthSmpls;
- WebRtc_Word8 _incomingPayload[MAX_INCOMING_PAYLOAD];
- WebRtc_UWord16 _payloadSizeBytes;
- WebRtc_UWord16 _realPayloadSizeBytes;
- WebRtc_Word32 _frequency;
- bool _firstTime;
- WebRtcRTPHeader _rtpInfo;
- WebRtc_UWord32 _nextTime;
+ private:
+ static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
+ WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
+ WebRtc_UWord32 ssrc);
+ RTPStream* _rtpStream;
+ WebRtc_Word32 _frequency;
+ WebRtc_Word16 _seqNo;
};
-class EncodeDecodeTest : public EncodeToFileTest
-{
-public:
- EncodeDecodeTest();
- EncodeDecodeTest(int testMode);
- virtual void Perform();
- WebRtc_UWord16 _playoutFreq;
- WebRtc_UWord8 _testMode;
-protected:
- Receiver _receiver;
+class Sender {
+ public:
+ Sender();
+ void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+ void Teardown();
+ void Run();
+ bool Add10MsData();
+ bool Process();
+
+ //for auto_test and logging
+ WebRtc_UWord8 testMode;
+ WebRtc_UWord8 codeId;
+
+ private:
+ AudioCodingModule* _acm;
+ PCMFile _pcmFile;
+ AudioFrame _audioFrame;
+ WebRtc_UWord16 _payloadSize;
+ WebRtc_UWord32 _timeStamp;
+ TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+ Receiver();
+ void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
+ void Teardown();
+ void Run();
+ bool IncomingPacket();
+ bool PlayoutData();
+
+ //for auto_test and logging
+ WebRtc_UWord8 codeId;
+ WebRtc_UWord8 testMode;
+
+ private:
+ AudioCodingModule* _acm;
+ bool _rtpEOF;
+ RTPStream* _rtpStream;
+ PCMFile _pcmFile;
+ WebRtc_Word16* _playoutBuffer;
+ WebRtc_UWord16 _playoutLengthSmpls;
+ WebRtc_Word8 _incomingPayload[MAX_INCOMING_PAYLOAD];
+ WebRtc_UWord16 _payloadSizeBytes;
+ WebRtc_UWord16 _realPayloadSizeBytes;
+ WebRtc_Word32 _frequency;
+ bool _firstTime;
+ WebRtcRTPHeader _rtpInfo;
+ WebRtc_UWord32 _nextTime;
+};
+
+class EncodeDecodeTest: public ACMTest {
+ public:
+ EncodeDecodeTest();
+ EncodeDecodeTest(int testMode);
+ virtual void Perform();
+
+ WebRtc_UWord16 _playoutFreq;
+ WebRtc_UWord8 _testMode;
+
+ private:
+ void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
+
+ protected:
+ Sender _sender;
+ Receiver _receiver;
};
-
+} // namespace webrtc
#endif
-
diff --git a/src/modules/audio_coding/main/test/EncodeToFileTest.cpp b/src/modules/audio_coding/main/test/EncodeToFileTest.cpp
deleted file mode 100644
index 01e7db1..0000000
--- a/src/modules/audio_coding/main/test/EncodeToFileTest.cpp
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2011 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 "EncodeToFileTest.h"
-
-#ifdef WIN32
-# include <Winsock2.h>
-#else
-# include <arpa/inet.h>
-#endif
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "gtest/gtest.h"
-
-TestPacketization::TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency)
-:
-_frequency(frequency),
-_seqNo(0)
-{
- _rtpStream = rtpStream;
-}
-
-TestPacketization::~TestPacketization()
-{
-}
-
-WebRtc_Word32 TestPacketization::SendData(
- const FrameType /* frameType */,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* /* fragmentation */)
-{
- _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, _frequency);
- //delete [] payloadData;
- return 1;
-}
-
-Sender::Sender()
-:
-_acm(NULL),
-//_payloadData(NULL),
-_payloadSize(0),
-_timeStamp(0)
-{
-}
-
-void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
-{
- acm->InitializeSender();
- struct CodecInst sendCodec;
- int noOfCodecs = acm->NumberOfCodecs();
- int codecNo;
-
- if (testMode == 1)
- {
- //set the codec, input file, and parameters for the current test
- codecNo = codeId;
- //use same input file for now
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- }
- else if (testMode == 0)
- {
- //set the codec, input file, and parameters for the current test
- codecNo = codeId;
- acm->Codec(codecNo, sendCodec);
- //use same input file for now
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- }
- else
- {
- printf("List of supported codec.\n");
- for(int n = 0; n < noOfCodecs; n++)
- {
- acm->Codec(n, sendCodec);
- printf("%d %s\n", n, sendCodec.plname);
- }
- printf("Choose your codec:");
- ASSERT_GT(scanf("%d", &codecNo), 0);
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- }
-
- acm->Codec(codecNo, sendCodec);
- acm->RegisterSendCodec(sendCodec);
- _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
- if(acm->RegisterTransportCallback(_packetization) < 0)
- {
- printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
- codeId);
- }
-
- _acm = acm;
-}
-
-void Sender::Teardown()
-{
- _pcmFile.Close();
- delete _packetization;
-}
-
-bool Sender::Add10MsData()
-{
- if (!_pcmFile.EndOfFile())
- {
- _pcmFile.Read10MsData(_audioFrame);
- WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
- if (ok != 0)
- {
- printf("Error calling Add10MsData: for run: codecId: %d\n",
- codeId);
- exit(1);
- }
- //_audioFrame._timeStamp += _pcmFile.PayloadLength10Ms();
- return true;
- }
- return false;
-}
-
-bool Sender::Process()
-{
- WebRtc_Word32 ok = _acm->Process();
- if (ok < 0)
- {
- printf("Error calling Add10MsData: for run: codecId: %d\n",
- codeId);
- exit(1);
- }
- return true;
-}
-
-void Sender::Run()
-{
- while (true)
- {
- if (!Add10MsData())
- {
- break;
- }
- if (!Process()) // This could be done in a processing thread
- {
- break;
- }
- }
-}
-
-EncodeToFileTest::EncodeToFileTest()
-{
-}
-
-
-void EncodeToFileTest::Perform(int fileType, int codeId, int* codePars, int testMode)
-{
- AudioCodingModule *acm = AudioCodingModule::Create(0);
- RTPFile rtpFile;
- char fileName[] = "outFile.rtp";
- rtpFile.Open(fileName, "wb+");
- rtpFile.WriteHeader();
-
- //for auto_test and logging
- _sender.testMode = testMode;
- _sender.codeId = codeId;
-
- _sender.Setup(acm, &rtpFile);
- struct CodecInst sendCodecInst;
- if(acm->SendCodec(sendCodecInst) >= 0)
- {
- _sender.Run();
- }
- _sender.Teardown();
- rtpFile.Close();
- AudioCodingModule::Destroy(acm);
-}
diff --git a/src/modules/audio_coding/main/test/EncodeToFileTest.h b/src/modules/audio_coding/main/test/EncodeToFileTest.h
deleted file mode 100644
index fdd3804..0000000
--- a/src/modules/audio_coding/main/test/EncodeToFileTest.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 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.
- */
-
-#ifndef ENCODETOFILETEST_H
-#define ENCODETOFILETEST_H
-
-#include "ACMTest.h"
-#include "audio_coding_module.h"
-#include "typedefs.h"
-#include "RTPFile.h"
-#include "PCMFile.h"
-#include <stdio.h>
-
-using namespace webrtc;
-
-// TestPacketization callback which writes the encoded payloads to file
-class TestPacketization : public AudioPacketizationCallback
-{
-public:
- TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
- ~TestPacketization();
- virtual WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
-private:
- static void MakeRTPheader(WebRtc_UWord8* rtpHeader,
- WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
- WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
- RTPStream* _rtpStream;
- WebRtc_Word32 _frequency;
- WebRtc_Word16 _seqNo;
-};
-
-class Sender
-{
-public:
- Sender();
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
- void Teardown();
- void Run();
- bool Add10MsData();
- bool Process();
-
- //for auto_test and logging
- WebRtc_UWord8 testMode;
- WebRtc_UWord8 codeId;
-
-private:
- AudioCodingModule* _acm;
- PCMFile _pcmFile;
- //WebRtc_Word16* _payloadData;
- AudioFrame _audioFrame;
- WebRtc_UWord16 _payloadSize;
- WebRtc_UWord32 _timeStamp;
- TestPacketization* _packetization;
-};
-
-// Test class
-class EncodeToFileTest : public ACMTest
-{
-public:
- EncodeToFileTest();
- virtual void Perform(int fileType, int codeId, int* codePars, int testMode);
-protected:
- Sender _sender;
-};
-
-#endif
diff --git a/src/modules/audio_coding/main/test/PCMFile.cpp b/src/modules/audio_coding/main/test/PCMFile.cpp
index e14101f..1942178 100644
--- a/src/modules/audio_coding/main/test/PCMFile.cpp
+++ b/src/modules/audio_coding/main/test/PCMFile.cpp
@@ -18,10 +18,10 @@
#include "gtest/gtest.h"
#include "module_common_types.h"
+namespace webrtc {
+
#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-
PCMFile::PCMFile():
_pcmFile(NULL),
_nSamples10Ms(160),
@@ -300,3 +300,5 @@
{
_readStereo = readStereo;
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/PCMFile.h b/src/modules/audio_coding/main/test/PCMFile.h
index dda02b7..5182a69 100644
--- a/src/modules/audio_coding/main/test/PCMFile.h
+++ b/src/modules/audio_coding/main/test/PCMFile.h
@@ -16,7 +16,7 @@
#include <cstdio>
#include <cstdlib>
-using namespace webrtc;
+namespace webrtc {
class PCMFile
{
@@ -60,4 +60,6 @@
bool _saveStereo;
};
+} // namespace webrtc
+
#endif
diff --git a/src/modules/audio_coding/main/test/RTPFile.cpp b/src/modules/audio_coding/main/test/RTPFile.cpp
index 04648be..b3eb5ce 100644
--- a/src/modules/audio_coding/main/test/RTPFile.cpp
+++ b/src/modules/audio_coding/main/test/RTPFile.cpp
@@ -20,9 +20,11 @@
#include "audio_coding_module.h"
#include "engine_configurations.h"
-#include "gtest/gtest.h"
+#include "gtest/gtest.h" // TODO (tlegrand): Consider removing usage of gtest.
#include "rw_lock_wrapper.h"
+namespace webrtc {
+
void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
{
rtpInfo->header.payloadType = rtpHeader[1];
@@ -123,21 +125,10 @@
}
else
{
- throw "Payload buffer too small";
- exit(1);
+ return -1;
}
-/*#ifdef WEBRTC_CODEC_G722
- if(ACMCodecDB::_mycodecs[ACMCodecDB::g722].pltype == packet->payloadType)
- {
- *offset = (packet->timeStamp/(packet->frequency/1000))<<1;
- }
- else
- {
-#endif*/
- *offset = (packet->timeStamp/(packet->frequency/1000));
-/*#ifdef WEBRTC_CODEC_G722
- }
-#endif*/
+ *offset = (packet->timeStamp/(packet->frequency/1000));
+
return packet->payloadSize;
}
@@ -189,15 +180,15 @@
WebRtc_UWord16 port, padding;
char fileHeader[40];
EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
- EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
+ EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
start_sec=ntohl(start_sec);
- EXPECT_GT(fread(&start_usec, 4, 1, _rtpFile), 0u);
+ EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
start_usec=ntohl(start_usec);
- EXPECT_GT(fread(&source, 4, 1, _rtpFile), 0u);
+ EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
source=ntohl(source);
- EXPECT_GT(fread(&port, 2, 1, _rtpFile), 0u);
+ EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
port=ntohs(port);
- EXPECT_GT(fread(&padding, 2, 1, _rtpFile), 0u);
+ EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
padding=ntohs(padding);
}
@@ -211,18 +202,8 @@
WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
WebRtc_UWord16 plen = htons(12 + payloadSize);
WebRtc_UWord32 offsetMs;
-/*#ifdef WEBRTC_CODEC_G722
- if(ACMCodecDB::_mycodecs[ACMCodecDB::g722].pltype == payloadType)
- {
- offsetMs = (timeStamp/(frequency/1000))<<1;
- }
- else
- {
-#endif*/
+
offsetMs = (timeStamp/(frequency/1000));
-/*#ifdef WEBRTC_CODEC_G722
- }
-#endif*/
offsetMs = htonl(offsetMs);
fwrite(&lengthBytes, 2, 1, _rtpFile);
fwrite(&plen, 2, 1, _rtpFile);
@@ -239,61 +220,41 @@
WebRtc_UWord16 lengthBytes;
WebRtc_UWord16 plen;
WebRtc_UWord8 rtpHeader[12];
- EXPECT_GT(fread(&lengthBytes, 2, 1, _rtpFile), 0u);
+ fread(&lengthBytes, 2, 1, _rtpFile);
+ /* Check if we have reached end of file. */
if (feof(_rtpFile))
{
_rtpEOF = true;
return 0;
}
- EXPECT_GT(fread(&plen, 2, 1, _rtpFile), 0u);
- if (feof(_rtpFile))
- {
- _rtpEOF = true;
- return 0;
- }
- EXPECT_GT(fread(offset, 4, 1, _rtpFile), 0u);
- if (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);
- if (plen < 12)
- {
- throw "Unable to read RTP file";
- exit(1);
- }
- EXPECT_GT(fread(rtpHeader, 12, 1, _rtpFile), 0u);
- if (feof(_rtpFile))
- {
- _rtpEOF = true;
- return 0;
- }
+ 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;
- if (lengthBytes != plen + 8)
- {
- throw "Length parameters in RTP file doesn't match";
- exit(1);
- }
+ EXPECT_EQ(lengthBytes, plen + 8);
+
if (plen == 0)
{
return 0;
}
- else if (lengthBytes - 20 > payloadSize)
+ if (payloadSize < (lengthBytes - 20))
{
- throw "Payload buffer too small";
- exit(1);
+ return -1;
}
lengthBytes -= 20;
- EXPECT_GT(fread(payloadData, 1, lengthBytes, _rtpFile), 0u);
- if (feof(_rtpFile))
+ if (lengthBytes < 0)
{
- _rtpEOF = true;
+ return -1;
}
+ EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
return lengthBytes;
}
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/RTPFile.h b/src/modules/audio_coding/main/test/RTPFile.h
index 23a43d0..db38a9f 100644
--- a/src/modules/audio_coding/main/test/RTPFile.h
+++ b/src/modules/audio_coding/main/test/RTPFile.h
@@ -18,7 +18,7 @@
#include <stdio.h>
#include <queue>
-using namespace webrtc;
+namespace webrtc {
class RTPStream
{
@@ -96,4 +96,5 @@
bool _rtpEOF;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/SpatialAudio.cpp b/src/modules/audio_coding/main/test/SpatialAudio.cpp
index 85c158f..b8c6856 100644
--- a/src/modules/audio_coding/main/test/SpatialAudio.cpp
+++ b/src/modules/audio_coding/main/test/SpatialAudio.cpp
@@ -18,7 +18,7 @@
#include "trace.h"
#include "common_types.h"
-using namespace webrtc;
+namespace webrtc {
#define NUM_PANN_COEFFS 10
@@ -236,4 +236,4 @@
_inFile.Rewind();
}
-
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/SpatialAudio.h b/src/modules/audio_coding/main/test/SpatialAudio.h
index 6a137d4..6a88327 100644
--- a/src/modules/audio_coding/main/test/SpatialAudio.h
+++ b/src/modules/audio_coding/main/test/SpatialAudio.h
@@ -19,6 +19,7 @@
#define MAX_FILE_NAME_LENGTH_BYTE 500
+namespace webrtc {
class SpatialAudio : public ACMTest
{
@@ -40,4 +41,7 @@
PCMFile _outFile;
int _testMode;
};
+
+} // namespace webrtc
+
#endif
diff --git a/src/modules/audio_coding/main/test/TestAllCodecs.cpp b/src/modules/audio_coding/main/test/TestAllCodecs.cpp
index 5f2c9d5..726112c 100644
--- a/src/modules/audio_coding/main/test/TestAllCodecs.cpp
+++ b/src/modules/audio_coding/main/test/TestAllCodecs.cpp
@@ -18,6 +18,8 @@
#include "trace.h"
#include "utility.h"
+namespace webrtc {
+
// Class for simulating packet handling
TestPack::TestPack():
_receiverACM(NULL),
@@ -114,7 +116,6 @@
_testMode = testMode;
}
-using namespace std;
TestAllCodecs::~TestAllCodecs()
{
if(_acmA != NULL)
@@ -143,7 +144,7 @@
if(_testMode == 0)
{
printf("Running All Codecs Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
"---------- TestAllCodecs ----------");
}
@@ -854,3 +855,5 @@
printf("%s\n", myCodecParam.plname);
}
+} // namespace webrtc
+
diff --git a/src/modules/audio_coding/main/test/TestAllCodecs.h b/src/modules/audio_coding/main/test/TestAllCodecs.h
index 958cefd..e0d621f 100644
--- a/src/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/src/modules/audio_coding/main/test/TestAllCodecs.h
@@ -15,6 +15,8 @@
#include "Channel.h"
#include "PCMFile.h"
+namespace webrtc {
+
class TestPack : public AudioPacketizationCallback
{
public:
@@ -89,6 +91,6 @@
int _counter;
};
-
#endif // TEST_ALL_CODECS_H
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/TestFEC.cpp b/src/modules/audio_coding/main/test/TestFEC.cpp
index ed61828..9376b75 100644
--- a/src/modules/audio_coding/main/test/TestFEC.cpp
+++ b/src/modules/audio_coding/main/test/TestFEC.cpp
@@ -19,6 +19,8 @@
#include "trace.h"
#include "utility.h"
+namespace webrtc {
+
TestFEC::TestFEC(int testMode):
_acmA(NULL),
_acmB(NULL),
@@ -28,8 +30,6 @@
_testMode = testMode;
}
-using namespace std;
-
TestFEC::~TestFEC()
{
if(_acmA != NULL)
@@ -55,7 +55,7 @@
if(_testMode == 0)
{
printf("Running FEC Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
"---------- TestFEC ----------");
}
char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
@@ -527,7 +527,7 @@
printf("Registering %s for side %c\n", codecName, side);
}
}
- cout << flush;
+ std::cout << std::flush;
AudioCodingModule* myACM;
switch(side)
{
@@ -619,3 +619,5 @@
_acmB->ReceiveCodec(myCodecParam);
printf("%s\n", myCodecParam.plname);
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/TestFEC.h b/src/modules/audio_coding/main/test/TestFEC.h
index 09d1009..00e951f 100644
--- a/src/modules/audio_coding/main/test/TestFEC.h
+++ b/src/modules/audio_coding/main/test/TestFEC.h
@@ -15,6 +15,8 @@
#include "Channel.h"
#include "PCMFile.h"
+namespace webrtc {
+
class TestFEC : public ACMTest
{
public:
@@ -42,6 +44,6 @@
int _testMode;
};
+} // namespace webrtc
#endif
-
diff --git a/src/modules/audio_coding/main/test/TestStereo.cpp b/src/modules/audio_coding/main/test/TestStereo.cpp
index c39d412..c38a07d 100644
--- a/src/modules/audio_coding/main/test/TestStereo.cpp
+++ b/src/modules/audio_coding/main/test/TestStereo.cpp
@@ -18,6 +18,7 @@
#include <cassert>
#include "trace.h"
+namespace webrtc {
// Class for simulating packet handling
TestPackStereo::TestPackStereo():
@@ -167,7 +168,6 @@
_testMode = testMode;
}
-using namespace std;
TestStereo::~TestStereo()
{
if(_acmA != NULL)
@@ -195,7 +195,7 @@
if(_testMode == 0)
{
printf("Running Stereo Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
"---------- TestStereo ----------");
}
@@ -550,3 +550,4 @@
}
}
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/TestStereo.h b/src/modules/audio_coding/main/test/TestStereo.h
index 07c32de..a447880 100644
--- a/src/modules/audio_coding/main/test/TestStereo.h
+++ b/src/modules/audio_coding/main/test/TestStereo.h
@@ -15,6 +15,8 @@
#include "Channel.h"
#include "PCMFile.h"
+namespace webrtc {
+
class TestPackStereo : public AudioPacketizationCallback
{
public:
@@ -94,6 +96,7 @@
int _codecType;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/TestVADDTX.cpp b/src/modules/audio_coding/main/test/TestVADDTX.cpp
index 3801653..6fd1ef8 100644
--- a/src/modules/audio_coding/main/test/TestVADDTX.cpp
+++ b/src/modules/audio_coding/main/test/TestVADDTX.cpp
@@ -17,6 +17,7 @@
#include <iostream>
#include "trace.h"
+namespace webrtc {
TestVADDTX::TestVADDTX(int testMode):
_acmA(NULL),
@@ -29,7 +30,6 @@
_testMode = testMode;
}
-using namespace std;
TestVADDTX::~TestVADDTX()
{
if(_acmA != NULL)
@@ -275,7 +275,7 @@
{
printf("Registering %s for side %c\n", codecName, side);
}
- cout << flush;
+ std::cout << std::flush;
AudioCodingModule* myACM;
switch(side)
{
@@ -500,3 +500,5 @@
getCounter[ii] = _counter[ii];
}
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/TestVADDTX.h b/src/modules/audio_coding/main/test/TestVADDTX.h
index cf9088b..e8f9e1e 100644
--- a/src/modules/audio_coding/main/test/TestVADDTX.h
+++ b/src/modules/audio_coding/main/test/TestVADDTX.h
@@ -15,6 +15,8 @@
#include "Channel.h"
#include "PCMFile.h"
+namespace webrtc {
+
typedef struct
{
bool statusDTX;
@@ -83,5 +85,6 @@
VADDTXstruct _getStruct;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/Tester.cpp b/src/modules/audio_coding/main/test/Tester.cpp
index 1049cad..c859c5f 100644
--- a/src/modules/audio_coding/main/test/Tester.cpp
+++ b/src/modules/audio_coding/main/test/Tester.cpp
@@ -16,7 +16,6 @@
#include "APITest.h"
#include "EncodeDecodeTest.h"
-#include "EncodeToFileTest.h"
#include "iSACTest.h"
#include "SpatialAudio.h"
#include "TestAllCodecs.h"
@@ -25,6 +24,9 @@
#include "TestVADDTX.h"
#include "TwoWayCommunication.h"
+using webrtc::AudioCodingModule;
+using webrtc::Trace;
+
// Be sure to create the following directories before running the tests:
// ./modules/audio_coding/main/test/res_tests
// ./modules/audio_coding/main/test/res_autotests
@@ -46,52 +48,52 @@
{
Trace::CreateTrace();
- Trace::SetTraceFile("./modules/audio_coding/main/test/res_tests/test_trace.txt");
+ Trace::SetTraceFile("acm_trace.txt");
printf("The following tests will be executed:\n");
#ifdef ACM_AUTO_TEST
printf(" ACM auto test\n");
- tests->push_back(new EncodeDecodeTest(0));
- tests->push_back(new TwoWayCommunication(0));
- tests->push_back(new TestAllCodecs(0));
- tests->push_back(new TestStereo(0));
- tests->push_back(new SpatialAudio(0));
- tests->push_back(new TestVADDTX(0));
- tests->push_back(new TestFEC(0));
- tests->push_back(new ISACTest(0));
+ tests->push_back(new webrtc::EncodeDecodeTest(0));
+ tests->push_back(new webrtc::TwoWayCommunication(0));
+ tests->push_back(new webrtc::TestAllCodecs(0));
+ tests->push_back(new webrtc::TestStereo(0));
+ tests->push_back(new webrtc::SpatialAudio(0));
+ tests->push_back(new webrtc::TestVADDTX(0));
+ tests->push_back(new webrtc::TestFEC(0));
+ tests->push_back(new webrtc::ISACTest(0));
#endif
#ifdef ACM_TEST_ENC_DEC
printf(" ACM encode-decode test\n");
- tests->push_back(new EncodeDecodeTest(2));
+ tests->push_back(new webrtc::EncodeDecodeTest(2));
#endif
#ifdef ACM_TEST_TWO_WAY
printf(" ACM two-way communication test\n");
- tests->push_back(new TwoWayCommunication(1));
+ tests->push_back(new webrtc::TwoWayCommunication(1));
#endif
#ifdef ACM_TEST_ALL_ENC_DEC
printf(" ACM all codecs test\n");
- tests->push_back(new TestAllCodecs(1));
+ tests->push_back(new webrtc::TestAllCodecs(1));
#endif
#ifdef ACM_TEST_STEREO
printf(" ACM stereo test\n");
- tests->push_back(new TestStereo(1));
- tests->push_back(new SpatialAudio(2));
+ tests->push_back(new webrtc::TestStereo(1));
+ tests->push_back(new webrtc::SpatialAudio(2));
#endif
#ifdef ACM_TEST_VAD_DTX
printf(" ACM VAD-DTX test\n");
- tests->push_back(new TestVADDTX(1));
+ tests->push_back(new webrtc::TestVADDTX(1));
#endif
#ifdef ACM_TEST_FEC
printf(" ACM FEC test\n");
- tests->push_back(new TestFEC(1));
+ tests->push_back(new webrtc::TestFEC(1));
#endif
#ifdef ACM_TEST_CODEC_SPEC_API
printf(" ACM codec API test\n");
- tests->push_back(new ISACTest(1));
+ tests->push_back(new webrtc::ISACTest(1));
#endif
#ifdef ACM_TEST_FULL_API
printf(" ACM full API test\n");
- tests->push_back(new APITest());
+ tests->push_back(new webrtc::APITest());
#endif
printf("\n");
}
diff --git a/src/modules/audio_coding/main/test/TwoWayCommunication.cpp b/src/modules/audio_coding/main/test/TwoWayCommunication.cpp
index bad71ab..e23c649 100644
--- a/src/modules/audio_coding/main/test/TwoWayCommunication.cpp
+++ b/src/modules/audio_coding/main/test/TwoWayCommunication.cpp
@@ -25,7 +25,7 @@
#include "trace.h"
#include "utility.h"
-using namespace webrtc;
+namespace webrtc {
#define MAX_FILE_NAME_LENGTH_BYTE 500
@@ -67,7 +67,8 @@
WebRtc_UWord8
-TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecID_B)
+TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
+ WebRtc_UWord8* codecID_B)
{
AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
WebRtc_UWord8 noCodec = tmpACM->NumberOfCodecs();
@@ -94,7 +95,8 @@
}
WebRtc_Word16
-TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen, WebRtc_UWord16* frequencyHz)
+TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen,
+ WebRtc_UWord16* frequencyHz)
{
WebRtc_Word8 tmpName[MAX_FILE_NAME_LENGTH_BYTE];
//strcpy(_fileName, "in.pcm");
@@ -139,7 +141,8 @@
{
strncpy(fileName, tmpName, len+1);
}
- printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequencyHz);
+ printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
+ *frequencyHz);
EXPECT_TRUE(fgets(tmpName, 6, stdin) != NULL);
WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
if(tmpFreq > 0)
@@ -174,7 +177,8 @@
CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
#ifdef WEBRTC_DTMF_DETECTION
_dtmfDetectorA = new(DTMFDetector);
- CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA, ACMUSA));
+ CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
+ ACMUSA));
#endif
//--- Set ref-A codecs
CHECK_ERROR(_acmRefA->RegisterSendCodec(codecInst_A));
@@ -185,7 +189,8 @@
CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
#ifdef WEBRTC_DTMF_DETECTION
_dtmfDetectorB = new(DTMFDetector);
- CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB, ACMUSA));
+ CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
+ ACMUSA));
#endif
//--- Set ref-B codecs
@@ -279,7 +284,8 @@
CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
#ifdef WEBRTC_DTMF_DETECTION
_dtmfDetectorA = new(DTMFDetector);
- CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA, ACMUSA));
+ CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
+ ACMUSA));
#endif
//--- Set ref-A codecs
@@ -291,7 +297,8 @@
CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
#ifdef WEBRTC_DTMF_DETECTION
_dtmfDetectorB = new(DTMFDetector);
- CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB, ACMUSA));
+ CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
+ ACMUSA));
#endif
//--- Set ref-B codecs
@@ -312,7 +319,8 @@
strcpy(fileName, "./src/modules/audio_coding/main/test/outAutotestA.pcm");
frequencyHz = 16000;
_outFileA.Open(fileName, frequencyHz, "wb");
- strcpy(refFileName, "./src/modules/audio_coding/main/test/ref_outAutotestA.pcm");
+ strcpy(refFileName,
+ "./src/modules/audio_coding/main/test/ref_outAutotestA.pcm");
_outFileRefA.Open(refFileName, frequencyHz, "wb");
//--- Input B
@@ -324,7 +332,8 @@
strcpy(fileName, "./src/modules/audio_coding/main/test/outAutotestB.pcm");
frequencyHz = 16000;
_outFileB.Open(fileName, frequencyHz, "wb");
- strcpy(refFileName, "./src/modules/audio_coding/main/test/ref_outAutotestB.pcm");
+ strcpy(refFileName,
+ "./src/modules/audio_coding/main/test/ref_outAutotestB.pcm");
_outFileRefB.Open(refFileName, frequencyHz, "wb");
//--- Set A-to-B channel
@@ -359,7 +368,8 @@
if(_testMode == 0)
{
printf("Running TwoWayCommunication Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- TwoWayCommunication ----------");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TwoWayCommunication ----------");
SetUpAutotest();
}
else
@@ -382,8 +392,8 @@
if(_testMode != 0)
{
printf("\n");
- printf("sec:msec A B\n");
- printf("-------- ----- -----\n");
+ printf("sec:msec A B\n");
+ printf("-------- ----- -----\n");
}
while(!_inFileA.EndOfFile() && !_inFileB.EndOfFile())
@@ -429,7 +439,8 @@
_acmA->ResetEncoder();
if(_testMode == 0)
{
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- Errors epected");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- Errors epected");
printf(".");
}
else
@@ -443,7 +454,8 @@
{
if(_testMode == 0)
{
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "----- END: Errors epected");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "----- END: Errors epected");
printf(".");
}
else
@@ -460,7 +472,8 @@
CHECK_ERROR(_acmB->ResetDecoder());
if(_testMode == 0)
{
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- Errors epected");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- Errors epected");
printf(".");
}
else
@@ -475,7 +488,8 @@
{
if(_testMode == 0)
{
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "----- END: Errors epected");
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "----- END: Errors epected");
printf(".");
}
else
@@ -500,6 +514,6 @@
_dtmfDetectorB->PrintDetectedDigits();
#endif
-
}
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/TwoWayCommunication.h b/src/modules/audio_coding/main/test/TwoWayCommunication.h
index 66ede04..0b33317 100644
--- a/src/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/src/modules/audio_coding/main/test/TwoWayCommunication.h
@@ -17,6 +17,7 @@
#include "audio_coding_module.h"
#include "utility.h"
+namespace webrtc {
class TwoWayCommunication : public ACMTest
{
@@ -58,5 +59,6 @@
int _testMode;
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/iSACTest.cpp b/src/modules/audio_coding/main/test/iSACTest.cpp
index 86271c2..bd066aa 100644
--- a/src/modules/audio_coding/main/test/iSACTest.cpp
+++ b/src/modules/audio_coding/main/test/iSACTest.cpp
@@ -28,6 +28,7 @@
#include "tick_util.h"
+namespace webrtc {
void SetISACConfigDefault(
ACMTestISACConfig& isacConfig)
@@ -595,3 +596,5 @@
_inFileA.Close();
_inFileB.Close();
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/iSACTest.h b/src/modules/audio_coding/main/test/iSACTest.h
index c6d4b9c..17bacad 100644
--- a/src/modules/audio_coding/main/test/iSACTest.h
+++ b/src/modules/audio_coding/main/test/iSACTest.h
@@ -21,6 +21,8 @@
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
+namespace webrtc {
+
struct ACMTestISACConfig
{
WebRtc_Word32 currentRateBitPerSec;
@@ -96,5 +98,6 @@
PCMFile _clientOutFile[NO_OF_CLIENTS];
};
+} // namespace webrtc
#endif
diff --git a/src/modules/audio_coding/main/test/utility.cpp b/src/modules/audio_coding/main/test/utility.cpp
index 5f83d96..b30f930 100644
--- a/src/modules/audio_coding/main/test/utility.cpp
+++ b/src/modules/audio_coding/main/test/utility.cpp
@@ -20,6 +20,7 @@
#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
+namespace webrtc {
ACMTestTimer::ACMTestTimer() :
_msec(0),
@@ -429,3 +430,5 @@
_numFrameTypes[frameType]++;
return 0;
}
+
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/test/utility.h b/src/modules/audio_coding/main/test/utility.h
index b25de44..651cfa6 100644
--- a/src/modules/audio_coding/main/test/utility.h
+++ b/src/modules/audio_coding/main/test/utility.h
@@ -13,6 +13,8 @@
#include "audio_coding_module.h"
+namespace webrtc {
+
//-----------------------------
#define CHECK_ERROR(f) \
do { \
@@ -88,8 +90,6 @@
} \
} while(0)
-using namespace webrtc;
-
class ACMTestTimer
{
public:
@@ -197,6 +197,6 @@
WebRtc_UWord32 _numFrameTypes[6];
};
-
+} // namespace webrtc
#endif // ACM_TEST_UTILITY_H
