commit 1cd46643d70c7d94fd42ef1d5346ac6e9f48dd5d
author flim <flim@webrtc.org> Tue Jul 26 09:01:39 2016 -0700
committer Commit bot <commit-bot@chromium.org> Tue Jul 26 16:01:45 2016 +0000
tree ee5b321c25056916a7651bd5d13c56e9512d5df7
parent 47bded4428ddc49ab75c5697e0b71cf7564e7e83

Disable some Opus tests pending an update

These tests will be reenabled and fixed after Opus 1.1.3 has landed in
Chromium and is rolled into WebRTC.

BUG=

Review-Url: https://codereview.webrtc.org/2185673002
Cr-Commit-Position: refs/heads/master@{#13534}

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 86f166b..8fce226 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -1418,7 +1418,7 @@
 }
 #endif
 
-TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "855041f2490b887302bce9d544731849",
@@ -1433,7 +1433,7 @@
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_Opus_stereo_20ms_voip) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   // If not set, default will be kAudio in case of stereo.
   EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
@@ -1621,7 +1621,7 @@
 #endif // WEBRTC_ANDROID
 }
 
-TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
+TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(100000, 32200, 51480);

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 99eb384..0510d70 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -492,7 +492,7 @@
 #else
 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
 #endif
-TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+TEST_F(NetEqDecodingTest, DISABLED_TestOpusBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");