Turning on Opus 120ms frame length switch.
Chromium has adopted Opus 1.2.1 which allows 120ms frame encoding. It
is time to turn on the switch for building WebRTC with this feature.
Bug: webrtc:8042
TBR: kjellander@webrtc.org
Change-Id: I644b47cfb56f835695ef1263741cda6e3ee3d862
Reviewed-on: https://chromium-review.googlesource.com/586725
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Felicia Lim <flim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19173}diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 9dfdb79..c16c5b1 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -1475,14 +1475,7 @@
}
#endif
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
-#define MAYBE_OpusFromFormat_stereo_20ms DISABLED_OpusFromFormat_stereo_20ms
-#else
-#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
-#define MAYBE_OpusFromFormat_stereo_20ms OpusFromFormat_stereo_20ms
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3e285b74510e62062fbd8142dacd16e9",
@@ -1518,15 +1511,7 @@
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
-#define MAYBE_OpusFromFormat_stereo_20ms_voip \
- DISABLED_OpusFromFormat_stereo_20ms_voip
-#else
-#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
-#define MAYBE_OpusFromFormat_stereo_20ms_voip OpusFromFormat_stereo_20ms_voip
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
// If not set, default will be kAudio in case of stereo.
EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
@@ -1545,7 +1530,7 @@
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
-TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms_voip) {
+TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
AudioEncoderOpus encoder(120, kOpusFormat);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120));
@@ -1646,15 +1631,7 @@
void Run(int expected_total_bits) { RunInner(expected_total_bits); }
};
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_48khz_20ms_10kbps DISABLED_Opus_48khz_20ms_10kbps
-#define MAYBE_OpusFromFormat_48khz_20ms_10kbps \
- DISABLED_OpusFromFormat_48khz_20ms_10kbps
-#else
-#define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
-#define MAYBE_OpusFromFormat_48khz_20ms_10kbps OpusFromFormat_48khz_20ms_10kbps
-#endif
-TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
+TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 8640);
@@ -1663,7 +1640,7 @@
#endif // WEBRTC_ANDROID
}
-TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) {
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
ASSERT_TRUE(SetUpSender());
@@ -1675,15 +1652,7 @@
#endif // WEBRTC_ANDROID
}
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_48khz_20ms_50kbps DISABLED_Opus_48khz_20ms_50kbps
-#define MAYBE_OpusFromFormat_48khz_20ms_50kbps \
- DISABLED_OpusFromFormat_48khz_20ms_50kbps
-#else
-#define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
-#define MAYBE_OpusFromFormat_48khz_20ms_50kbps OpusFromFormat_48khz_20ms_50kbps
-#endif
-TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
+TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 45792);
@@ -1692,7 +1661,7 @@
#endif // WEBRTC_ANDROID
}
-TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) {
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
ASSERT_TRUE(SetUpSender());
@@ -1706,7 +1675,7 @@
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
-#if defined(WEBRTC_ANDROID) || WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#if defined(WEBRTC_ANDROID)
#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
DISABLED_OpusFromFormat_48khz_20ms_100kbps
@@ -1789,12 +1758,7 @@
uint32_t frame_size_samples_;
};
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_48khz_20ms_10kbps_2 DISABLED_Opus_48khz_20ms_10kbps
-#else
-#define MAYBE_Opus_48khz_20ms_10kbps_2 Opus_48khz_20ms_10kbps
-#endif
-TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps_2) {
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(10000, 29512, 4800);
@@ -1803,12 +1767,7 @@
#endif // WEBRTC_ANDROID
}
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_48khz_20ms_50kbps_2 DISABLED_Opus_48khz_20ms_50kbps
-#else
-#define MAYBE_Opus_48khz_20ms_50kbps_2 Opus_48khz_20ms_50kbps
-#endif
-TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps_2) {
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
Run(50000, 29512, 23304);
@@ -1817,13 +1776,7 @@
#endif // WEBRTC_ANDROID
}
-
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-#define MAYBE_Opus_48khz_20ms_100kbps_2 DISABLED_Opus_48khz_20ms_100kbps
-#else
-#define MAYBE_Opus_48khz_20ms_100kbps_2 Opus_48khz_20ms_100kbps
-#endif
-TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps_2) {
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps_2) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID)
#if defined(WEBRTC_ARCH_ARM64)
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 6879058..66f6537 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -465,8 +465,7 @@
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
- defined(WEBRTC_CODEC_OPUS) && \
- !WEBRTC_OPUS_SUPPORT_120MS_PTIME
+ defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
diff --git a/webrtc/webrtc.gni b/webrtc/webrtc.gni
index 676ab2f..e18464f 100644
--- a/webrtc/webrtc.gni
+++ b/webrtc/webrtc.gni
@@ -38,7 +38,7 @@
# Enable this if the Opus version upon which WebRTC is built supports direct
# encoding of 120 ms packets.
- rtc_opus_support_120ms_ptime = false
+ rtc_opus_support_120ms_ptime = true
# Enable this to let the Opus audio codec change complexity on the fly.
rtc_opus_variable_complexity = false