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/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 =