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Disable some Opus tests pending an update

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These tests will be reenabled and updated after Opus has been updated in
Chromium and rolled into WebRTC.

BUG=737323, webrtc:8024

Review-Url: https://codereview.webrtc.org/2963673002
Cr-Commit-Position: refs/heads/master@{#19118}
This commit is contained in:
flim 2017-07-24 02:17:38 -07:00 committed by Commit Bot
parent f3a48ab6dc
commit bf8202185c
2 changed files with 14 additions and 14 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

26
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -1482,7 +1482,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
#define MAYBE_OpusFromFormat_stereo_20ms OpusFromFormat_stereo_20ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_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",
@ -1499,7 +1499,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms) {
const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
AudioEncoderOpus encoder(120, kOpusFormat);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120));
@ -1526,7 +1526,7 @@ TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
#define MAYBE_OpusFromFormat_stereo_20ms_voip OpusFromFormat_stereo_20ms_voip
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_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));
@ -1545,7 +1545,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms_voip) {
TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms_voip) {
const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
AudioEncoderOpus encoder(120, kOpusFormat);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120));
@ -1654,7 +1654,7 @@ class AcmSetBitRateNewApi : public AcmSetBitRateTest {
#define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
#define MAYBE_OpusFromFormat_48khz_20ms_10kbps OpusFromFormat_48khz_20ms_10kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 9288);
@ -1663,7 +1663,7 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) {
TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_10kbps) {
AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
ASSERT_TRUE(SetUpSender());
@ -1683,7 +1683,7 @@ TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) {
#define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
#define MAYBE_OpusFromFormat_48khz_20ms_50kbps OpusFromFormat_48khz_20ms_50kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 47960);
@ -1692,7 +1692,7 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) {
TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_50kbps) {
AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
ASSERT_TRUE(SetUpSender());
@ -1715,12 +1715,12 @@ TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) {
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
OpusFromFormat_48khz_20ms_100kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
TEST_F(AcmSetBitRateOldApi, DISABLED_Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
Run(100000, 100888);
}
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_100kbps) {
AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
ASSERT_TRUE(SetUpSender());
@ -1794,7 +1794,7 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
#else
#define MAYBE_Opus_48khz_20ms_10kbps_2 Opus_48khz_20ms_10kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps_2) {
TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_10kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 32200, 5176);
@ -1808,7 +1808,7 @@ TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps_2) {
#else
#define MAYBE_Opus_48khz_20ms_50kbps_2 Opus_48khz_20ms_50kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps_2) {
TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_50kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 32200, 24768);
@ -1823,7 +1823,7 @@ TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps_2) {
#else
#define MAYBE_Opus_48khz_20ms_100kbps_2 Opus_48khz_20ms_100kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps_2) {
TEST_F(AcmChangeBitRateOldApi, DISABLED_Opus_48khz_20ms_100kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
#if defined(WEBRTC_ARCH_ARM64)

2
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -471,7 +471,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
#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");