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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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