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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc | 20 |
1 files changed, 9 insertions, 11 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc index 77bd5b5035..aec7e580ec 100644 --- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc +++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc @@ -31,7 +31,6 @@ #include "modules/include/module_common_types_public.h" #include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/ignore_wundef.h" #include "rtc_base/message_digest.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/strings/string_builder.h" @@ -77,11 +76,11 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); const std::string output_checksum = - "fec6827bb9ee0b21770bbbb4a3a6f8823bf537dc|" - "3610cc7be4b3407b9c273b1299ab7f8f47cca96b"; + "2efdbea92c3fb2383c59f89d881efec9f94001d0|" + "a6831b946b59913852ae3e53f99fa8f209bb23cd"; const std::string network_stats_checksum = - "3d043e47e5f4bb81d37e7bce8c44bf802965c853|" + "dfaf4399fd60293405290476ccf1c05c807c71a0|" "076662525572dba753b11578330bd491923f7f5e"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, @@ -99,11 +98,11 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) { webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp"); const std::string output_checksum = - "b3c4899eab5378ef5e54f2302948872149f6ad5e|" - "589e975ec31ea13f302457fea1425be9380ffb96"; + "7eddce841cbfa500964c91cdae78b01b9f448948|" + "5d13affec87bf4cc8c7667f0cd0d25e1ad09c7c3"; const std::string network_stats_checksum = - "dc8447b9fee1a21fd5d1f4045d62b982a3fb0215"; + "92b0fdcbf8bb9354d40140b7312f2fb76a078555"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, absl::GetFlag(FLAGS_gen_ref)); @@ -165,7 +164,7 @@ TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) { const double kDriftFactor = 1000.0 / (1000.0 + 25.0); const double kNetworkFreezeTimeMs = 0.0; const bool kGetAudioDuringFreezeRecovery = false; - const int kDelayToleranceMs = 20; + const int kDelayToleranceMs = 60; const int kMaxTimeToSpeechMs = 100; LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs, kGetAudioDuringFreezeRecovery, kDelayToleranceMs, @@ -495,7 +494,7 @@ TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { timestamp += kCngPeriodSamples; uint32_t first_speech_timestamp = timestamp; // Insert speech again. - for (int i = 0; i < 3; ++i) { + for (int i = 0; i < 4; ++i) { PopulateRtpInfo(seq_no, timestamp, &rtp_info); ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload)); ++seq_no; @@ -700,8 +699,7 @@ TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) { for (int i = 0; i < 5; ++i) { InsertPacket(kSamples * (i - 1000)); } - EXPECT_FALSE(GetAudioReturnMuted()); - EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_); + GetAudioUntilNormal(); } // Verifies that NetEq doesn't enter muted state when CNG mode is active and the |