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: https://chromium.googlesource.com/chromium/src/+log/a8e5140..c6076f2
Full diff: https://chromium.googlesource.com/chromium/src/+/a8e5140..c6076f2
Changed dependencies:
* src/third_party/catapult: https://chromium.googlesource.com/external/github.com/catapult-project/catapult.git/+log/471db30..d4d48e6
* src/third_party/opus/src: https://chromium.googlesource.com/chromium/deps/opus.git/+log/cae6961..655cc54
DEPS diff: https://chromium.googlesource.com/chromium/src/+/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}
diff --git a/DEPS b/DEPS
index ce12c35..e164282 100644
--- a/DEPS
+++ b/DEPS
@@ -6,7 +6,7 @@
vars = {
'extra_gyp_flag': '-Dextra_gyp_flag=0',
'chromium_git': 'https://chromium.googlesource.com',
- 'chromium_revision': 'a8e5140aa654117793cdd6654248202045900384',
+ 'chromium_revision': 'c6076f2213725d472f17191c54ce0e3232a07537',
}
# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
diff --git a/resources/audio_coding/neteq4_opus_network_stats.dat.sha1 b/resources/audio_coding/neteq4_opus_network_stats.dat.sha1
index 6a9e7ee..e19ee80 100644
--- a/resources/audio_coding/neteq4_opus_network_stats.dat.sha1
+++ b/resources/audio_coding/neteq4_opus_network_stats.dat.sha1
@@ -1 +1 @@
-cc9fa62d0a8f46ffebc782aea2610dda67bb5558
\ No newline at end of file
+dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref.pcm.sha1
index 5cecc50..3c4e897 100644
--- a/resources/audio_coding/neteq4_opus_ref.pcm.sha1
+++ b/resources/audio_coding/neteq4_opus_ref.pcm.sha1
@@ -1 +1 @@
-301895f1aaa9cd9eae0f5d04d179d63491d744cc
\ No newline at end of file
+c23004d91ffbe5e7a1f24620fc89b58c0426040f
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1
index b7cf990..3c4e897 100644
--- a/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1
+++ b/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1
@@ -1 +1 @@
-fbad99878c7a26958e755190027c976692708334
\ No newline at end of file
+c23004d91ffbe5e7a1f24620fc89b58c0426040f
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1
index b7cf990..3c4e897 100644
--- a/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1
+++ b/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1
@@ -1 +1 @@
-fbad99878c7a26958e755190027c976692708334
\ No newline at end of file
+c23004d91ffbe5e7a1f24620fc89b58c0426040f
\ No newline at end of file
diff --git a/resources/short_mixed_mono_48.dat.sha1 b/resources/short_mixed_mono_48.dat.sha1
index efc5745..afee167 100644
--- a/resources/short_mixed_mono_48.dat.sha1
+++ b/resources/short_mixed_mono_48.dat.sha1
@@ -1 +1 @@
-6d3d824f7f73e1ab852982a7b53d52cfcbbf72cc
\ No newline at end of file
+8c18538d80ea6583dbed0fe5cdabb1ea030ae72b
\ No newline at end of file
diff --git a/resources/short_mixed_mono_48_arm.dat.sha1 b/resources/short_mixed_mono_48_arm.dat.sha1
new file mode 100644
index 0000000..917c351
--- /dev/null
+++ b/resources/short_mixed_mono_48_arm.dat.sha1
@@ -0,0 +1 @@
+7973eb3a87109c6cdb133e7c4a47d7e70b1782ba
\ No newline at end of file
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 384db86..67a44bd 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
@@ -1406,13 +1406,13 @@
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"855041f2490b887302bce9d544731849",
"855041f2490b887302bce9d544731849",
- "1e1a0fce893fef2d66886a7f09e2ebce",
- "7417a66c28be42d5d9b2d64e0c191585"),
+ "9692eede45638eb425e0daf9c75b5c7a",
+ "c4faa472fbb0730370aaf34920381a09"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"d781cce1ab986b618d0da87226cdde30",
"d781cce1ab986b618d0da87226cdde30",
- "1a1fe04dd12e755949987c8d729fb3e0",
- "47b0b04f1d03076b857c86c72c2c298b"),
+ "8d6782b905c3230d4b0e3e83e1fc3439",
+ "8b0126eab82d9e4e367ab33ded2f1a8e"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
@@ -1423,13 +1423,13 @@
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"9b9e12bc3cc793740966e11cbfa8b35b",
"9b9e12bc3cc793740966e11cbfa8b35b",
- "57412a4b5771d19ff03ec35deffe7067",
- "7ad0bbefcaa87e23187bf4a56d2f3513"),
+ "0de6249018fdd316c21086db84e10610",
+ "fd21a19b6b1e891f5daea6c4a299c254"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"c7340b1189652ab6b5e80dade7390cb4",
"c7340b1189652ab6b5e80dade7390cb4",
- "cdfe85939c411d12b61701c566e22d26",
- "7a678fbe46df5bf0c67e88264a2d9275"),
+ "95612864c954ee63e28cc6eebad56626",
+ "49954b0d5a5f705a8798e7071b0c6f36"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
@@ -1497,9 +1497,9 @@
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
- Run(10000, 9328);
+ Run(10000, 9288);
#else
- Run(10000, 9072);
+ Run(10000, 9024);
#endif // WEBRTC_ANDROID
}
@@ -1507,9 +1507,9 @@
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
- Run(50000, 47952);
+ Run(50000, 47960);
#else
- Run(50000, 49600);
+ Run(50000, 49544);
#endif // WEBRTC_ANDROID
}
@@ -1589,18 +1589,18 @@
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
- Run(10000, 32200, 5496);
+ Run(10000, 32200, 5176);
#else
- Run(10000, 32200, 5432);
+ Run(10000, 32200, 5456);
#endif // WEBRTC_ANDROID
}
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
- Run(50000, 32200, 24912);
+ Run(50000, 32200, 24768);
#else
- Run(50000, 32200, 24792);
+ Run(50000, 32200, 24848);
#endif // WEBRTC_ANDROID
}
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index c82b184..b6112d1 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -206,9 +206,9 @@
const int kCheckTimeMs = 1500;
#if defined(OPUS_FIXED_POINT)
- const uint16_t kOutputValueBound = 20;
+ const uint16_t kOutputValueBound = 30;
#else
- const uint16_t kOutputValueBound = 2;
+ const uint16_t kOutputValueBound = 8;
#endif
int time = 0;
diff --git a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
index e4db3a3..371282c 100644
--- a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
@@ -61,9 +61,15 @@
}
TEST(AudioClassifierTest, DoAnalysisMono) {
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+ RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
+ test::ResourcePath("short_mixed_mono_48_arm", "dat"),
+ 1);
+#else
RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
test::ResourcePath("short_mixed_mono_48", "dat"),
1);
+#endif // WEBRTC_ARCH_ARM
}
TEST(AudioClassifierTest, DoAnalysisStereo) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index f218f72..a304e82 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -568,8 +568,12 @@
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string input_ref_file =
+ // The pcm files were generated by using Opus v1.1.2 to decode the RTC
+ // file generated by Opus v1.1
webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
const std::string network_stat_ref_file =
+ // The network stats file was generated when using Opus v1.1.2 to decode
+ // the RTC file generated by Opus v1.1
webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
"dat");
const std::string rtcp_stat_ref_file =
diff --git a/webrtc/modules/modules_unittests.isolate b/webrtc/modules/modules_unittests.isolate
index 10eb6f4..cbef944 100644
--- a/webrtc/modules/modules_unittests.isolate
+++ b/webrtc/modules/modules_unittests.isolate
@@ -114,6 +114,7 @@
'<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin',
'<(DEPTH)/resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin',
'<(DEPTH)/resources/short_mixed_mono_48.dat',
+ '<(DEPTH)/resources/short_mixed_mono_48_arm.dat',
'<(DEPTH)/resources/short_mixed_mono_48.pcm',
'<(DEPTH)/resources/short_mixed_stereo_48.dat',
'<(DEPTH)/resources/short_mixed_stereo_48.pcm',
