Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2
Includes updates to tests for Opus v.1.1.2, reveiwed in https://codereview.webrtc.org/1629413002/ Change log:a8e5140..c6076f2Full diff:a8e5140..c6076f2Changed dependencies: * src/third_party/catapult:471db30..d4d48e6* src/third_party/opus/src:cae6961..655cc54DEPS diff:a8e5140..c6076f2/DEPS No update to Clang. BUG=chromium:580524 TBR= Review URL: https://codereview.webrtc.org/1657343002 Cr-Commit-Position: refs/heads/master@{#11464}
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@ -1406,13 +1406,13 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
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Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
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"855041f2490b887302bce9d544731849",
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"855041f2490b887302bce9d544731849",
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"1e1a0fce893fef2d66886a7f09e2ebce",
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"7417a66c28be42d5d9b2d64e0c191585"),
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"9692eede45638eb425e0daf9c75b5c7a",
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"c4faa472fbb0730370aaf34920381a09"),
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AcmReceiverBitExactnessOldApi::PlatformChecksum(
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"d781cce1ab986b618d0da87226cdde30",
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"d781cce1ab986b618d0da87226cdde30",
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"1a1fe04dd12e755949987c8d729fb3e0",
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"47b0b04f1d03076b857c86c72c2c298b"),
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"8d6782b905c3230d4b0e3e83e1fc3439",
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"8b0126eab82d9e4e367ab33ded2f1a8e"),
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50, test::AcmReceiveTestOldApi::kStereoOutput);
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}
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@ -1423,13 +1423,13 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
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Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
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"9b9e12bc3cc793740966e11cbfa8b35b",
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"9b9e12bc3cc793740966e11cbfa8b35b",
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"57412a4b5771d19ff03ec35deffe7067",
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"7ad0bbefcaa87e23187bf4a56d2f3513"),
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"0de6249018fdd316c21086db84e10610",
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"fd21a19b6b1e891f5daea6c4a299c254"),
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AcmReceiverBitExactnessOldApi::PlatformChecksum(
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"c7340b1189652ab6b5e80dade7390cb4",
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"c7340b1189652ab6b5e80dade7390cb4",
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"cdfe85939c411d12b61701c566e22d26",
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"7a678fbe46df5bf0c67e88264a2d9275"),
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"95612864c954ee63e28cc6eebad56626",
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"49954b0d5a5f705a8798e7071b0c6f36"),
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50, test::AcmReceiveTestOldApi::kStereoOutput);
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}
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@ -1497,9 +1497,9 @@ class AcmSetBitRateOldApi : public ::testing::Test {
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TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
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ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
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#if defined(WEBRTC_ANDROID)
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Run(10000, 9328);
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Run(10000, 9288);
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#else
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Run(10000, 9072);
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Run(10000, 9024);
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#endif // WEBRTC_ANDROID
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}
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@ -1507,9 +1507,9 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
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TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
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ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
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#if defined(WEBRTC_ANDROID)
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Run(50000, 47952);
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Run(50000, 47960);
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#else
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Run(50000, 49600);
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Run(50000, 49544);
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#endif // WEBRTC_ANDROID
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}
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@ -1589,18 +1589,18 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
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TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
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ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
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#if defined(WEBRTC_ANDROID)
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Run(10000, 32200, 5496);
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Run(10000, 32200, 5176);
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#else
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Run(10000, 32200, 5432);
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Run(10000, 32200, 5456);
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#endif // WEBRTC_ANDROID
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}
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TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
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ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
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#if defined(WEBRTC_ANDROID)
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Run(50000, 32200, 24912);
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Run(50000, 32200, 24768);
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#else
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Run(50000, 32200, 24792);
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Run(50000, 32200, 24848);
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#endif // WEBRTC_ANDROID
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}
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@ -206,9 +206,9 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
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const int kCheckTimeMs = 1500;
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#if defined(OPUS_FIXED_POINT)
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const uint16_t kOutputValueBound = 20;
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const uint16_t kOutputValueBound = 30;
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#else
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const uint16_t kOutputValueBound = 2;
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const uint16_t kOutputValueBound = 8;
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#endif
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int time = 0;
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@ -61,9 +61,15 @@ void RunAnalysisTest(const std::string& audio_filename,
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}
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TEST(AudioClassifierTest, DoAnalysisMono) {
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#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
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RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
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test::ResourcePath("short_mixed_mono_48_arm", "dat"),
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1);
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#else
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RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
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test::ResourcePath("short_mixed_mono_48", "dat"),
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1);
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#endif // WEBRTC_ARCH_ARM
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}
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TEST(AudioClassifierTest, DoAnalysisStereo) {
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@ -568,8 +568,12 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
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const std::string input_rtp_file =
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webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
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const std::string input_ref_file =
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// The pcm files were generated by using Opus v1.1.2 to decode the RTC
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// file generated by Opus v1.1
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webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
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const std::string network_stat_ref_file =
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// The network stats file was generated when using Opus v1.1.2 to decode
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// the RTC file generated by Opus v1.1
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webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
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"dat");
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const std::string rtcp_stat_ref_file =
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