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Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Issue 2528993002: Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (Closed)
Patch Set: Created 4 years ago
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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(&timestamp)); 286 EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
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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