Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics
The two tests both read and process the same (rather long) RTP input
file, and simply look at different outputs. This change merges the two
tests into one, in order to reduce testing time.
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6865 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 317c7b5..1fd5345 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -200,11 +200,10 @@
void LoadDecoders();
void OpenInputFile(const std::string &rtp_file);
void Process(NETEQTEST_RTPpacket* rtp_ptr, int* out_len);
- void DecodeAndCompare(const std::string &rtp_file,
- const std::string &ref_file);
- void DecodeAndCheckStats(const std::string &rtp_file,
- const std::string &stat_ref_file,
- const std::string &rtcp_ref_file);
+ void DecodeAndCompare(const std::string& rtp_file,
+ const std::string& ref_file,
+ const std::string& stat_ref_file,
+ const std::string& rtcp_ref_file);
static void PopulateRtpInfo(int frame_index,
int timestamp,
WebRtcRTPHeader* rtp_info);
@@ -337,8 +336,10 @@
sim_clock_ += kTimeStepMs;
}
-void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
- const std::string &ref_file) {
+void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
+ const std::string& ref_file,
+ const std::string& stat_ref_file,
+ const std::string& rtcp_ref_file) {
OpenInputFile(rtp_file);
std::string ref_out_file = "";
@@ -347,6 +348,18 @@
}
RefFiles ref_files(ref_file, ref_out_file);
+ std::string stat_out_file = "";
+ if (stat_ref_file.empty()) {
+ stat_out_file = webrtc::test::OutputPath() + "neteq_network_stats.dat";
+ }
+ RefFiles network_stat_files(stat_ref_file, stat_out_file);
+
+ std::string rtcp_out_file = "";
+ if (rtcp_ref_file.empty()) {
+ rtcp_out_file = webrtc::test::OutputPath() + "neteq_rtcp_stats.dat";
+ }
+ RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
+
NETEQTEST_RTPpacket rtp;
ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
int i = 0;
@@ -357,32 +370,6 @@
int out_len = 0;
ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
- }
-}
-
-void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file,
- const std::string &stat_ref_file,
- const std::string &rtcp_ref_file) {
- OpenInputFile(rtp_file);
- std::string stat_out_file = "";
- if (stat_ref_file.empty()) {
- stat_out_file = webrtc::test::OutputPath() +
- "neteq_network_stats.dat";
- }
- RefFiles network_stat_files(stat_ref_file, stat_out_file);
-
- std::string rtcp_out_file = "";
- if (rtcp_ref_file.empty()) {
- rtcp_out_file = webrtc::test::OutputPath() +
- "neteq_rtcp_stats.dat";
- }
- RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
-
- NETEQTEST_RTPpacket rtp;
- ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
- while (rtp.dataLen() >= 0) {
- int out_len;
- ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
@@ -432,17 +419,6 @@
// have a copy of the file, the test will fail.
const std::string input_ref_file =
webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
-
- if (FLAGS_gen_ref) {
- DecodeAndCompare(input_rtp_file, "");
- } else {
- DecodeAndCompare(input_rtp_file, input_ref_file);
- }
-}
-
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
- const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
- "resources/audio_coding/neteq_universal_new.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
// For Visual Studio 2012 and later, we will have to use the generic reference
// file, rather than the windows-specific one.
@@ -454,11 +430,14 @@
#endif
const std::string rtcp_stat_ref_file =
webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
+
if (FLAGS_gen_ref) {
- DecodeAndCheckStats(input_rtp_file, "", "");
+ DecodeAndCompare(input_rtp_file, "", "", "");
} else {
- DecodeAndCheckStats(input_rtp_file, network_stat_ref_file,
- rtcp_stat_ref_file);
+ DecodeAndCompare(input_rtp_file,
+ input_ref_file,
+ network_stat_ref_file,
+ rtcp_stat_ref_file);
}
}
