Formatting ACM tests
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
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 58e6299..949507d 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -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 @@
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 @@
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 @@
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 @@
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::Teardown() {
- delete [] _playoutBuffer;
+ delete[] _playoutBuffer;
_pcmFile.Close();
if (testMode > 1)
Trace::ReturnTrace();
@@ -205,16 +203,16 @@
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 @@
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 @@
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 @@
}
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 @@
AudioCodingModule::Destroy(acm);
}
-} // namespace webrtc
+} // namespace webrtc