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Skipping some Opus tests to let the new roll flow.

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In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].

They will be re-enabled and fixed as soon as the new Opus revision is
rolled.

[1] - https://chromium-review.googlesource.com/1061499

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
This commit is contained in:
Mirko Bonadei
2018-05-18 08:33:05 +02:00
committed by Commit Bot
parent 4c8811b255
commit 638edfc88c
1 changed files with 18 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

24
modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -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);