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Cleaning up neteq_unittest

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- Conforming to testing standards.
- Fixing a way of generating new reference output files.
- ifdef the test to run only on linux 64-bit
- Renaming unittest source file.
- Renaming test vectors

Review URL: http://webrtc-codereview.appspot.com/296007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1028 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2011-11-25 13:43:42 +00:00
parent 789da89d37
commit 0fcc2eb368
2 changed files with 60 additions and 31 deletions
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3
src/modules/audio_coding/neteq/neteq.gypi
View File

@ -95,9 +95,10 @@
'NetEq',
'NetEqTestTools',
'<(webrtc_root)/../testing/gtest.gyp:gtest',
'<(webrtc_root)/../test/test.gyp:test_support_main',
],
'sources': [
'neteq_api_unittest.cc',
'webrtc_neteq_unittest.cc',
],
}, # neteq_unittests
{

88
src/modules/audio_coding/neteq/neteq_api_unittest.cc → src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
View File

@ -16,6 +16,7 @@
#include <stdlib.h>
#include <string.h> // memset
#include <string>
#include <vector>
#include "gtest/gtest.h"
@ -26,8 +27,9 @@
#include "typedefs.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/interface/webrtc_neteq.h"
#include "modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
#include "testsupport/fileutils.h"
namespace {
namespace webrtc {
class NetEqDecodingTest : public ::testing::Test {
protected:
@ -36,7 +38,8 @@ class NetEqDecodingTest : public ::testing::Test {
virtual void TearDown();
void SelectDecoders(WebRtcNetEQDecoder* used_codec);
void LoadDecoders();
void DecodeAndCompare(const char* rtp_file, const char* ref_file);
void DecodeAndCompare(const std::string &rtp_file,
const std::string &ref_file);
NETEQTEST_NetEQClass* neteq_inst_;
std::vector<NETEQTEST_Decoder*> dec_;
@ -90,19 +93,33 @@ void NetEqDecodingTest::LoadDecoders() {
}
}
void NetEqDecodingTest::DecodeAndCompare(const char* rtp_file,
const char* ref_file) {
void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
const std::string &ref_file) {
NETEQTEST_RTPpacket rtp;
FILE* rtp_fp = fopen(rtp_file, "rb");
FILE* rtp_fp = fopen(rtp_file.c_str(), "rb");
ASSERT_TRUE(rtp_fp != NULL);
ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp));
ASSERT_GT(rtp.readFromFile(rtp_fp), 0);
FILE* ref_fp = fopen(ref_file, "rb");
ASSERT_TRUE(ref_fp != NULL);
FILE* ref_fp = NULL;
FILE* out_fp = NULL;
if (!ref_file.empty()) {
ref_fp = fopen(ref_file.c_str(), "rb");
ASSERT_TRUE(ref_fp != NULL);
} else {
std::string out_file = webrtc::test::OutputPath() + "neteq_out.pcm";
out_fp = fopen(out_file.c_str(), "wb");
ASSERT_TRUE(out_fp != NULL);
}
unsigned int sim_clock = 0;
const int kTimeStep = 10;
// NetEQ must be polled for data once every 10 ms. Thus, neither of the
// constants below can be changed.
const int kTimeStepMs = 10;
const int kBlockSize8kHz = kTimeStepMs * 8;
const int kBlockSize16kHz = kTimeStepMs * 16;
const int kBlockSize32kHz = kTimeStepMs * 32;
const int kMaxBlockSize = kBlockSize32kHz;
while (rtp.dataLen() >= 0) {
// Check if time to receive.
while ((sim_clock >= rtp.time()) &&
@ -115,38 +132,49 @@ void NetEqDecodingTest::DecodeAndCompare(const char* rtp_file,
}
// RecOut
WebRtc_Word16 out_data[10 * 32]; // 10 ms at 32 kHz
WebRtc_Word16 out_data[kMaxBlockSize];
WebRtc_Word16 out_len = neteq_inst_->recOut(out_data);
ASSERT_TRUE((out_len == 80) || (out_len == 160) || (out_len == 320));
ASSERT_TRUE((out_len == kBlockSize8kHz) ||
(out_len == kBlockSize16kHz) ||
(out_len == kBlockSize32kHz));
// Read from ref file
WebRtc_Word16 ref_data[10 * 32]; // 10 ms at 32 kHz
if (static_cast<size_t>(out_len) !=
fread(ref_data, sizeof(WebRtc_Word16), out_len, ref_fp)) {
break;
if (ref_fp) {
// Read from ref file.
WebRtc_Word16 ref_data[kMaxBlockSize];
if (static_cast<size_t>(out_len) !=
fread(ref_data, sizeof(WebRtc_Word16), out_len, ref_fp)) {
break;
}
// Compare
EXPECT_EQ(0, memcmp(out_data, ref_data, sizeof(WebRtc_Word16) * out_len));
}
// Compare
EXPECT_EQ(0, memcmp(out_data, ref_data, sizeof(WebRtc_Word16) * out_len));
if (out_fp) {
// Write to output file (mainly for generating new output vectors).
ASSERT_EQ(static_cast<size_t>(out_len),
fwrite(out_data, sizeof(WebRtc_Word16), out_len, out_fp));
}
// Increase time
sim_clock += kTimeStep;
// Increase time.
sim_clock += kTimeStepMs;
}
ASSERT_NE(0, feof(ref_fp)); // Make sure that we reached the end.
fclose(rtp_fp);
fclose(ref_fp);
if (ref_fp) {
ASSERT_NE(0, feof(ref_fp)); // Make sure that we reached the end.
fclose(ref_fp);
}
if (out_fp) fclose(out_fp);
}
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_64_BITS)
TEST_F(NetEqDecodingTest, TestBitExactness) {
DecodeAndCompare("test/data/audio_coding/universal.rtp",
"test/data/audio_coding/universal_ref.pcm");
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"test/data/audio_coding/universal.rtp";
const std::string kInputRefFile = webrtc::test::ProjectRootPath() +
"test/data/audio_coding/universal_ref.pcm";
DecodeAndCompare(kInputRtpFile, kInputRefFile);
}
#endif // defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_64_BITS)
} // namespace
int main(int argc, char** argv) {
::testing::InitGoogleTest(&argc, argv);
return RUN_ALL_TESTS();
}