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Turning on Opus 120ms frame length switch.

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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}
This commit is contained in:
minyue-webrtc
2017-07-27 17:45:49 +02:00
committed by Commit Bot
parent 28e2919cfd
commit 516711cde9
3 changed files with 13 additions and 61 deletions
Download Patch File Download Diff File Expand all files Collapse all files

69
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -1475,14 +1475,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
} }
#endif #endif
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"3e285b74510e62062fbd8142dacd16e9", "3e285b74510e62062fbd8142dacd16e9",
@ -1518,15 +1511,7 @@ TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
50, test::AcmReceiveTestOldApi::kStereoOutput); 50, test::AcmReceiveTestOldApi::kStereoOutput);
} }
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
// If not set, default will be kAudio in case of stereo. // If not set, default will be kAudio in case of stereo.
EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip)); EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
@ -1545,7 +1530,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
50, test::AcmReceiveTestOldApi::kStereoOutput); 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"}}); const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
AudioEncoderOpus encoder(120, kOpusFormat); AudioEncoderOpus encoder(120, kOpusFormat);
ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120)); ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120));
@ -1646,15 +1631,7 @@ class AcmSetBitRateNewApi : public AcmSetBitRateTest {
void Run(int expected_total_bits) { RunInner(expected_total_bits); } void Run(int expected_total_bits) { RunInner(expected_total_bits); }
}; };
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
Run(10000, 8640); Run(10000, 8640);
@ -1663,7 +1640,7 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
} }
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) { TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
AudioEncoderOpus encoder( AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}})); 107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
ASSERT_TRUE(SetUpSender()); ASSERT_TRUE(SetUpSender());
@ -1675,15 +1652,7 @@ TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) {
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
} }
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
Run(50000, 45792); Run(50000, 45792);
@ -1692,7 +1661,7 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
} }
TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) { TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
AudioEncoderOpus encoder( AudioEncoderOpus encoder(
107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}})); 107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
ASSERT_TRUE(SetUpSender()); ASSERT_TRUE(SetUpSender());
@ -1706,7 +1675,7 @@ TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) {
// The result on the Android platforms is inconsistent for this test case. // 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. // 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_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \ #define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
DISABLED_OpusFromFormat_48khz_20ms_100kbps DISABLED_OpusFromFormat_48khz_20ms_100kbps
@ -1789,12 +1758,7 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
uint32_t frame_size_samples_; uint32_t frame_size_samples_;
}; };
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
Run(10000, 29512, 4800); Run(10000, 29512, 4800);
@ -1803,12 +1767,7 @@ TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps_2) {
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
} }
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
Run(50000, 29512, 23304); Run(50000, 29512, 23304);
@ -1817,13 +1776,7 @@ TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps_2) {
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
} }
TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps_2) {
#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) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
#if defined(WEBRTC_ARCH_ARM64) #if defined(WEBRTC_ARCH_ARM64)

3
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -465,8 +465,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS) && \ defined(WEBRTC_CODEC_OPUS)
!WEBRTC_OPUS_SUPPORT_120MS_PTIME
#define MAYBE_TestOpusBitExactness TestOpusBitExactness #define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else #else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness

2
webrtc/webrtc.gni
View File

@ -38,7 +38,7 @@ declare_args() {
# Enable this if the Opus version upon which WebRTC is built supports direct # Enable this if the Opus version upon which WebRTC is built supports direct
# encoding of 120 ms packets. # 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. # Enable this to let the Opus audio codec change complexity on the fly.
rtc_opus_variable_complexity = false rtc_opus_variable_complexity = false