Fixing a few tests for the upcoming Opus 1.2-alpha.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1439 matching lines @@
           "7190ee718ab3d80eca181e5f7140c210", "android_arm32_audio",
           "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "66516152eeaa1e650ad94ff85f668dac",
           "66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload",
           "android_arm64_payload"),
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 #endif
 
-TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
+#else
+#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
+#endif
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "855041f2490b887302bce9d544731849",
           "855041f2490b887302bce9d544731849",
           "9692eede45638eb425e0daf9c75b5c7a",
           "86d3552bb3492247f965cdd0e88a1c82"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "d781cce1ab986b618d0da87226cdde30",
           "d781cce1ab986b618d0da87226cdde30",
           "8d6782b905c3230d4b0e3e83e1fc3439",
           "798347a685fac7d0c2d8f748ffe66881"),
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
+#else
+#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
+#endif
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_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));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "9b9e12bc3cc793740966e11cbfa8b35b",
           "9b9e12bc3cc793740966e11cbfa8b35b",
           "0de6249018fdd316c21086db84e10610",
           "9c4cb69db77b85841a5f8225bb8f508b"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "c7340b1189652ab6b5e80dade7390cb4",
@@ skipping 57 matching lines @@
     ASSERT_TRUE(SetUpSender());
     ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
                                   payload_type, codec_frame_size_samples,
                                   codec_frame_size_rtp_timestamps));
   }
 
   std::unique_ptr<test::AcmSendTestOldApi> send_test_;
   std::unique_ptr<test::InputAudioFile> audio_source_;
 };
 
-TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_48khz_20ms_10kbps DISABLED_Opus_48khz_20ms_10kbps
+#else
+#define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
+#endif
+TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(10000, 9288);
 #else
   Run(10000, 9024);
 #endif  // WEBRTC_ANDROID
 }
 
-TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_48khz_20ms_50kbps DISABLED_Opus_48khz_20ms_50kbps
+#else
+#define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
+#endif
+TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(50000, 47960);
 #else
   Run(50000, 49544);
 #endif  // WEBRTC_ANDROID
 }
 
 // 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)
+#if defined(WEBRTC_ANDROID) || WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
 #else
 #define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
 #endif
 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
   Run(100000, 100888);
 }
 
 // These next 2 tests ensure that the SetBitRate function has no effect on PCM
@@ skipping 50 matching lines @@
       packet_counter++;
     }
     EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
     EXPECT_EQ(expected_after_switch_bits, nr_bytes_after * 8);
   }
 
   uint32_t sampling_freq_hz_;
   uint32_t frame_size_samples_;
 };
 
-TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_48khz_20ms_10kbps DISABLED_Opus_48khz_20ms_10kbps
+#else
+#define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
+#endif
+TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {

[minyue-webrtc, 2017/03/14 14:18:14]

note that MAYBE_Opus_48khz_20ms_10kbps is redefined. I hope it is ok.

[kjellander (google.com), 2017/03/14 14:36:47]

Run a few trybots to find out. I think redefinitions are considered compile
errors.

   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(10000, 32200, 5176);
 #else
   Run(10000, 32200, 5456);
 #endif  // WEBRTC_ANDROID
 }
 
-TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_48khz_20ms_50kbps DISABLED_Opus_48khz_20ms_50kbps
+#else
+#define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
+#endif
+TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(50000, 32200, 24768);
 #else
   Run(50000, 32200, 24848);
 #endif  // WEBRTC_ANDROID
 }
 
-TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
+
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
+#else
+#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
+#endif
+TEST_F(AcmChangeBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   #if defined(WEBRTC_ARCH_ARM64)
     Run(100000, 32200, 51152);
   #else
     Run(100000, 32200, 51248);
   #endif  // WEBRTC_ARCH_ARM64
 #else
   Run(100000, 32200, 50584);
 #endif  // WEBRTC_ANDROID
@@ skipping 166 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc