Disable opus tests to allow upgrade to opus 1.3
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.
Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30425}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index fb26025..febb121 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -1455,13 +1455,15 @@
"27fef7b799393347ec3b5694369a1c36");
} // namespace
-TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSenderBitExactnessOldApi, DISABLED_Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(audio_checksum, payload_checksum, 50,
test::AcmReceiveTestOldApi::kStereoOutput);
}
-TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
@@ -1518,7 +1520,8 @@
50, test::AcmReceiveTestOldApi::kQuadOutput, decoder_factory);
}
-TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms_voip) {
auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
// If not set, default will be kAudio in case of stereo.
@@ -1618,7 +1621,8 @@
}
};
-TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_10kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
ASSERT_TRUE(SetUpSender());
@@ -1627,7 +1631,8 @@
RunInner(8000, 12000);
}
-TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_50kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
ASSERT_TRUE(SetUpSender());
@@ -1735,7 +1740,8 @@
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
OpusFromFormat_48khz_20ms_100kbps
#endif
-TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_100kbps) {
const auto config = AudioEncoderOpus::SdpToConfig(
SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
ASSERT_TRUE(SetUpSender());
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 698b441..2192f40 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -912,7 +912,8 @@
EXPECT_EQ(64000, config.bitrate_bps);
}
-TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_P(AudioEncoderOpusTest, DISABLED_OpusFlagDtxAsNonSpeech) {
// Create encoder with DTX enabled.
AudioEncoderOpusConfig config;
config.dtx_enabled = true;
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 2c8edf0..be209e4 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -101,7 +101,8 @@
} // namespace
-TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST(BandwidthAdaptationTest, DISABLED_BandwidthAdaptationTest) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-AdjustOpusBandwidth/Enabled/");
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 0788601..c3281b4 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -599,7 +599,8 @@
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
}
-TEST_P(AudioDecoderOpusTest, EncodeDecode) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_P(AudioDecoderOpusTest, DISABLED_EncodeDecode) {
constexpr int tolerance = 6176;
const int channel_diff_tolerance = opus_sample_rate_hz_ == 16000 ? 6 : 0;
constexpr double mse = 238630.0;
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index e284ddb..9a53fdd 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -105,7 +105,8 @@
#else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
#endif
-TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(NetEqDecodingTest, DISABLED_TestOpusBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
@@ -134,7 +135,8 @@
#else
#define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
#endif
-TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
+// TODO(webrtc:11325) Reenable after Opus has been upgraded to 1.3.
+TEST_F(NetEqDecodingTest, DISABLED_TestOpusDtxBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");