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Unified Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

Issue 2158293003: Update tests and DTX check for Opus 1.1.3. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 months ago
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Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 86f166bcf07938c649f8bbb154e4e8a7b228bf71..5412c9a1a9419a07d14ca159a6b8166cfb0dc90d 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -1424,12 +1424,12 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
"855041f2490b887302bce9d544731849",
"855041f2490b887302bce9d544731849",
"9692eede45638eb425e0daf9c75b5c7a",
- "c4faa472fbb0730370aaf34920381a09"),
+ "86d3552bb3492247f965cdd0e88a1c82"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"d781cce1ab986b618d0da87226cdde30",
"d781cce1ab986b618d0da87226cdde30",
"8d6782b905c3230d4b0e3e83e1fc3439",
- "8b0126eab82d9e4e367ab33ded2f1a8e"),
+ "798347a685fac7d0c2d8f748ffe66881"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
@@ -1441,12 +1441,12 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
"9b9e12bc3cc793740966e11cbfa8b35b",
"9b9e12bc3cc793740966e11cbfa8b35b",
"0de6249018fdd316c21086db84e10610",
- "fd21a19b6b1e891f5daea6c4a299c254"),
+ "9c4cb69db77b85841a5f8225bb8f508b"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"c7340b1189652ab6b5e80dade7390cb4",
"c7340b1189652ab6b5e80dade7390cb4",
"95612864c954ee63e28cc6eebad56626",
- "49954b0d5a5f705a8798e7071b0c6f36"),
+ "ae33ea2e43407cf9ebdabbbd6ca912a3"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
@@ -1624,7 +1624,7 @@ TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
- Run(100000, 32200, 51480);
+ Run(100000, 32200, 51248);
#else
Run(100000, 32200, 50584);
#endif // WEBRTC_ANDROID

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