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Fixing a few tests for the upcoming Opus 1.2-alpha.

Browse Source 
BUG=b/35415318

NOTRY=True

Review-Url: https://codereview.webrtc.org/2746763005
Cr-Commit-Position: refs/heads/master@{#17234}
This commit is contained in:
minyue
2017-03-14 14:33:30 -07:00
committed by Commit bot
parent 275e2099ab
commit a613eb6bff
3 changed files with 61 additions and 12 deletions
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52
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -1457,7 +1457,12 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
}
#endif
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
#else
#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"855041f2490b887302bce9d544731849",
@ -1472,7 +1477,12 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
#else
#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_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));
@ -1550,7 +1560,12 @@ class AcmSetBitRateOldApi : public ::testing::Test {
std::unique_ptr<test::InputAudioFile> audio_source_;
};
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_10kbps DISABLED_Opus_48khz_20ms_10kbps
#else
#define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 9288);
@ -1559,7 +1574,12 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
#endif // WEBRTC_ANDROID
}
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_50kbps DISABLED_Opus_48khz_20ms_50kbps
#else
#define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
#endif
TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 47960);
@ -1570,7 +1590,7 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
#if defined(WEBRTC_ANDROID)
#if defined(WEBRTC_ANDROID) || WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
#else
#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
@ -1641,7 +1661,12 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
uint32_t frame_size_samples_;
};
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_10kbps_2 DISABLED_Opus_48khz_20ms_10kbps
#else
#define MAYBE_Opus_48khz_20ms_10kbps_2 Opus_48khz_20ms_10kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 32200, 5176);
@ -1650,7 +1675,12 @@ TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_50kbps_2 DISABLED_Opus_48khz_20ms_50kbps
#else
#define MAYBE_Opus_48khz_20ms_50kbps_2 Opus_48khz_20ms_50kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 32200, 24768);
@ -1659,7 +1689,13 @@ TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_Opus_48khz_20ms_100kbps_2 DISABLED_Opus_48khz_20ms_100kbps
#else
#define MAYBE_Opus_48khz_20ms_100kbps_2 Opus_48khz_20ms_100kbps
#endif
TEST_F(AcmChangeBitRateOldApi, MAYBE_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)

18
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -644,7 +644,7 @@ TEST_P(OpusTest, OpusDurationEstimation) {
}
TEST_P(OpusTest, OpusDecodeRepacketized) {
const int kPackets = 6;
constexpr size_t kPackets = 6;
PrepareSpeechData(channels_, 20, 20 * kPackets);
@ -668,14 +668,26 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
new int16_t[kPackets * kOpus20msFrameSamples * channels_]);
OpusRepacketizer* rp = opus_repacketizer_create();
for (int idx = 0; idx < kPackets; idx++) {
size_t num_packets = 0;
constexpr size_t kMaxCycles = 100;
for (size_t idx = 0; idx < kMaxCycles; ++idx) {
auto speech_block = speech_data_.GetNextBlock();
encoded_bytes_ =
WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
if (opus_repacketizer_cat(rp, bitstream_, encoded_bytes_) == OPUS_OK) {
++num_packets;
if (num_packets == kPackets) {
break;
}
} else {
// Opus repacketizer cannot guarantee a success. We try again if it fails.
opus_repacketizer_init(rp);
num_packets = 0;
}
}
EXPECT_EQ(kPackets, num_packets);
encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);

3
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -468,7 +468,8 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
defined(WEBRTC_CODEC_OPUS) && \
!WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness