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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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97 EXPECT_EQ(17U, red_->Max10MsFramesInAPacket()); | 97 EXPECT_EQ(17U, red_->Max10MsFramesInAPacket()); |
98 } | 98 } |
99 | 99 |
100 TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) { | 100 TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) { |
101 EXPECT_CALL(*mock_encoder_, | 101 EXPECT_CALL(*mock_encoder_, |
102 OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>())); | 102 OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>())); |
103 red_->OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()); | 103 red_->OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()); |
104 } | 104 } |
105 | 105 |
106 TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) { | 106 TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) { |
107 EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5)); | 107 EXPECT_CALL(*mock_encoder_, |
108 red_->OnReceivedUplinkPacketLossFraction(0.5); | 108 OnReceivedUplinkPacketLossFraction(rtc::Optional<float>(0.5f))); |
| 109 red_->OnReceivedUplinkPacketLossFraction(rtc::Optional<float>(0.5f)); |
109 } | 110 } |
110 | 111 |
111 // Checks that the an Encode() call is immediately propagated to the speech | 112 // Checks that the an Encode() call is immediately propagated to the speech |
112 // encoder. | 113 // encoder. |
113 TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) { | 114 TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) { |
114 // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction | 115 // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction |
115 // check ensures that exactly one call to EncodeImpl happens in each | 116 // check ensures that exactly one call to EncodeImpl happens in each |
116 // Encode call. | 117 // Encode call. |
117 InSequence s; | 118 InSequence s; |
118 MockFunction<void(int check_point_id)> check; | 119 MockFunction<void(int check_point_id)> check; |
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298 config.speech_encoder = NULL; | 299 config.speech_encoder = NULL; |
299 EXPECT_DEATH(red = new AudioEncoderCopyRed(std::move(config)), | 300 EXPECT_DEATH(red = new AudioEncoderCopyRed(std::move(config)), |
300 "Speech encoder not provided."); | 301 "Speech encoder not provided."); |
301 // The delete operation is needed to avoid leak reports from memcheck. | 302 // The delete operation is needed to avoid leak reports from memcheck. |
302 delete red; | 303 delete red; |
303 } | 304 } |
304 | 305 |
305 #endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) | 306 #endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
306 | 307 |
307 } // namespace webrtc | 308 } // namespace webrtc |
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