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Remove support for old test modes in EncodeDecodeTest

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This test is so old, it used to be interactive with an automated mode
bolted on to the side. That automatic mode is the only one that's used
nowadays.

Bug: webrtc:8396
Change-Id: I3b473f53ff6afa363b9691e8471a5754f46d3d3f
Reviewed-on: https://webrtc-review.googlesource.com/83583
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23620}
This commit is contained in:
Karl Wiberg
2018-06-14 13:12:05 +02:00
committed by Commit Bot
parent d477129ac0
commit 88aee288f8
3 changed files with 23 additions and 73 deletions
Download Patch File Download Diff File Expand all files Collapse all files

52
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -54,7 +54,6 @@ Sender::Sender()
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
@ -69,19 +68,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
_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));
@ -150,21 +137,8 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
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];
@ -249,15 +223,10 @@ void Receiver::Run() {
}
}
EncodeDecodeTest::EncodeDecodeTest() {
_testMode = 2;
}
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;
EncodeDecodeTest::EncodeDecodeTest(int test_mode) {
// There used to be different test modes. The only one still supported is the
// "autotest" mode.
RTC_CHECK_EQ(0, test_mode);
}
void EncodeDecodeTest::Perform() {
@ -275,7 +244,6 @@ void EncodeDecodeTest::Perform() {
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) {
@ -290,19 +258,13 @@ void EncodeDecodeTest::Perform() {
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);
std::string fileName = EncodeToFile(1, codeId, codePars);
RTPFile rtpFile;
rtpFile.Open(fileName.c_str(), "rb");
@ -320,8 +282,7 @@ void EncodeDecodeTest::Perform() {
std::string EncodeDecodeTest::EncodeToFile(int fileType,
int codeId,
int* codePars,
int testMode) {
int* codePars) {
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
RTPFile rtpFile;
@ -331,7 +292,6 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
rtpFile.WriteHeader();
// Store for auto_test and logging.
_sender.testMode = testMode;
_sender.codeId = codeId;
_sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);

12
modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -54,8 +54,6 @@ class Sender {
void Run();
bool Add10MsData();
//for auto_test and logging
uint8_t testMode;
uint8_t codeId;
protected:
@ -80,7 +78,6 @@ class Receiver {
//for auto_test and logging
uint8_t codeId;
uint8_t testMode;
private:
PCMFile _pcmFile;
@ -101,18 +98,13 @@ class Receiver {
class EncodeDecodeTest : public ACMTest {
public:
EncodeDecodeTest();
explicit EncodeDecodeTest(int testMode);
explicit EncodeDecodeTest(int test_mode);
void Perform() override;
uint16_t _playoutFreq;
uint8_t _testMode;
private:
std::string EncodeToFile(int fileType,
int codeId,
int* codePars,
int testMode);
std::string EncodeToFile(int fileType, int codeId, int* codePars);
protected:
Sender _sender;

2
modules/audio_coding/test/PacketLossTest.cc
View File

@ -142,7 +142,6 @@ void PacketLossTest::Perform() {
rtpFile.Open(fileName.c_str(), "wb+");
rtpFile.WriteHeader();
sender_->testMode = 0;
sender_->codeId = codec_id;
sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
@ -157,7 +156,6 @@ void PacketLossTest::Perform() {
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
receiver_->testMode = 0;
receiver_->codeId = codec_id;
receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,