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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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| 94 test_ssrc = 3456; | 94 test_ssrc = 3456; |
| 95 test_timestamp = 4567; | 95 test_timestamp = 4567; |
| 96 test_sequence_number = 2345; | 96 test_sequence_number = 2345; |
| 97 } | 97 } |
| 98 ~RtpRtcpAudioTest() {} | 98 ~RtpRtcpAudioTest() {} |
| 99 | 99 |
| 100 void SetUp() override { | 100 void SetUp() override { |
| 101 receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock)); | 101 receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock)); |
| 102 receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock)); | 102 receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock)); |
| 103 | 103 |
| 104 rtp_payload_registry1_.reset(new RTPPayloadRegistry()); | 104 rtp_payload_registry1_.reset(new RTPPayloadRegistry( |
| 105 rtp_payload_registry2_.reset(new RTPPayloadRegistry()); | 105 RTPPayloadStrategy::CreateStrategy(true))); |
| 106 rtp_payload_registry2_.reset(new RTPPayloadRegistry( |
| 107 RTPPayloadStrategy::CreateStrategy(true))); |
| 106 | 108 |
| 107 RtpRtcp::Configuration configuration; | 109 RtpRtcp::Configuration configuration; |
| 108 configuration.audio = true; | 110 configuration.audio = true; |
| 109 configuration.clock = &fake_clock; | 111 configuration.clock = &fake_clock; |
| 110 configuration.receive_statistics = receive_statistics1_.get(); | 112 configuration.receive_statistics = receive_statistics1_.get(); |
| 111 configuration.outgoing_transport = &transport1; | 113 configuration.outgoing_transport = &transport1; |
| 112 configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; | 114 configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; |
| 113 | 115 |
| 114 module1.reset(RtpRtcp::CreateRtpRtcp(configuration)); | 116 module1.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
| 115 rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver( | 117 rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver( |
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| 281 nullptr, nullptr, nullptr)); | 283 nullptr, nullptr, nullptr)); |
| 282 | 284 |
| 283 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC()); | 285 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC()); |
| 284 EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp)); | 286 EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp)); |
| 285 EXPECT_EQ(test_timestamp + in_timestamp, timestamp); | 287 EXPECT_EQ(test_timestamp + in_timestamp, timestamp); |
| 286 in_timestamp += 10; | 288 in_timestamp += 10; |
| 287 } | 289 } |
| 288 } | 290 } |
| 289 | 291 |
| 290 } // namespace webrtc | 292 } // namespace webrtc |
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