From 0fcc2eb3688dd31c40eb3d458ca41a9d54a6d82c Mon Sep 17 00:00:00 2001 From: "henrik.lundin@webrtc.org" Date: Fri, 25 Nov 2011 13:43:42 +0000 Subject: [PATCH] Cleaning up neteq_unittest - 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 --- src/modules/audio_coding/neteq/neteq.gypi | 3 +- ...i_unittest.cc => webrtc_neteq_unittest.cc} | 88 ++++++++++++------- 2 files changed, 60 insertions(+), 31 deletions(-) rename src/modules/audio_coding/neteq/{neteq_api_unittest.cc => webrtc_neteq_unittest.cc} (59%) diff --git a/src/modules/audio_coding/neteq/neteq.gypi b/src/modules/audio_coding/neteq/neteq.gypi index 261c0e5f1..c72efdd1a 100644 --- a/src/modules/audio_coding/neteq/neteq.gypi +++ b/src/modules/audio_coding/neteq/neteq.gypi @@ -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 { diff --git a/src/modules/audio_coding/neteq/neteq_api_unittest.cc b/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc similarity index 59% rename from src/modules/audio_coding/neteq/neteq_api_unittest.cc rename to src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc index 1e757e0e5..b6b1082e6 100644 --- a/src/modules/audio_coding/neteq/neteq_api_unittest.cc +++ b/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc @@ -16,6 +16,7 @@ #include #include // memset +#include #include #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 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(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(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(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(); -} -