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Formatting ACM tests

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Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/

Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/1342004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-05-03 07:34:12 +00:00
parent 03efc89151
commit d5726a1286
34 changed files with 4385 additions and 5230 deletions
Download Patch File Download Diff File Expand all files Collapse all files

75
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@ -27,20 +27,18 @@
namespace webrtc {
TestPacketization::TestPacketization(RTPStream *rtpStream,
uint16_t frequency)
TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
: _rtpStream(rtpStream),
_frequency(frequency),
_seqNo(0) {
}
TestPacketization::~TestPacketization() { }
TestPacketization::~TestPacketization() {
}
int32_t TestPacketization::SendData(
const FrameType /* frameType */,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const FrameType /* frameType */, const uint8_t payloadType,
const uint32_t timeStamp, const uint8_t* payloadData,
const uint16_t payloadSize,
const RTPFragmentationHeader* /* fragmentation */) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
@ -62,8 +60,8 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
int codecNo;
// Open input file
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
const std::string file_name = webrtc::test::ResourcePath(
"audio_coding/testfile32kHz", "pcm");
_pcmFile.Open(file_name, 32000, "rb");
// Set the codec for the current test.
@ -127,7 +125,7 @@ void Sender::Run() {
if (!Add10MsData()) {
break;
}
if (!Process()) { // This could be done in a processing thread
if (!Process()) { // This could be done in a processing thread
break;
}
}
@ -155,16 +153,16 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
int playSampFreq;
std::string file_name;
std::stringstream file_stream;
file_stream << webrtc::test::OutputPath() << "encodeDecode_out" <<
static_cast<int>(codeId) << ".pcm";
file_stream << webrtc::test::OutputPath() << "encodeDecode_out"
<< static_cast<int>(codeId) << ".pcm";
file_name = file_stream.str();
_rtpStream = rtpStream;
if (testMode == 1) {
playSampFreq=recvCodec.plfreq;
playSampFreq = recvCodec.plfreq;
_pcmFile.Open(file_name, recvCodec.plfreq, "wb+");
} else if (testMode == 0) {
playSampFreq=32000;
playSampFreq = 32000;
_pcmFile.Open(file_name, 32000, "wb+");
} else {
printf("\nValid output frequencies:\n");
@ -172,7 +170,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
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() + "encodeDecode_out.pcm";
file_name = webrtc::test::OutputPath() + "encodeDecode_out.pcm";
_pcmFile.Open(file_name, playSampFreq, "wb+");
}
@ -184,7 +182,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
}
void Receiver::Teardown() {
delete [] _playoutBuffer;
delete[] _playoutBuffer;
_pcmFile.Close();
if (testMode > 1)
Trace::ReturnTrace();
@ -205,16 +203,16 @@ bool Receiver::IncomingPacket() {
return false;
}
}
}
}
int32_t ok = _acm->IncomingPacket(_incomingPayload,
_realPayloadSizeBytes, _rtpInfo);
if (ok != 0) {
printf("Error when inserting packet to ACM, for run: codecId: %d\n",
codeId);
}
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_payloadSizeBytes, &_nextTime);
int32_t ok = _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
_rtpInfo);
if (ok != 0) {
printf("Error when inserting packet to ACM, for run: codecId: %d\n",
codeId);
}
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
_firstTime = true;
}
@ -233,8 +231,7 @@ bool Receiver::PlayoutData() {
if (_playoutLengthSmpls == 0) {
return false;
}
_pcmFile.Write10MsData(audioFrame.data_,
audioFrame.samples_per_channel_);
_pcmFile.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
return true;
}
@ -265,20 +262,20 @@ void Receiver::Run() {
EncodeDecodeTest::EncodeDecodeTest() {
_testMode = 2;
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_encdec_trace.txt").c_str());
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());
}
_testMode = testMode;
if (_testMode != 0) {
Trace::CreateTrace();
Trace::SetTraceFile(
(webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
}
}
void EncodeDecodeTest::Perform() {
@ -289,9 +286,9 @@ 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.
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;
@ -390,4 +387,4 @@ void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
AudioCodingModule::Destroy(acm);
}
} // namespace webrtc
} // namespace webrtc