diff options
| -rw-r--r-- | modules/audio_coding/acm2/audio_coding_module_unittest.cc | 24 |
1 files changed, 18 insertions, 6 deletions
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index ff5431dba9..57471ec106 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -1634,7 +1634,9 @@ class AcmSetBitRateNewApi : public AcmSetBitRateTest { void Run(int expected_total_bits) { RunInner(expected_total_bits); } }; -TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +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, 8640); @@ -1643,7 +1645,9 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) { #endif // WEBRTC_ANDROID } -TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_10kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}})); const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107); @@ -1656,7 +1660,9 @@ TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) { #endif // WEBRTC_ANDROID } -TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +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, 45792); @@ -1665,7 +1671,9 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) { #endif // WEBRTC_ANDROID } -TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +TEST_F(AcmSetBitRateNewApi, DISABLED_OpusFromFormat_48khz_20ms_50kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}})); const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107); @@ -1766,7 +1774,9 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi { uint32_t frame_size_samples_; }; -TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +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, 29512, 4800); @@ -1775,7 +1785,9 @@ TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) { #endif // WEBRTC_ANDROID } -TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) { +// TODO(bugs.webrtc.org/9280): Re-enable when the new Opus revision is rolled +// into WebRTC. +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, 29512, 23304); |