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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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| 47 receiver_ = new TestRtpReceiver(); | 47 receiver_ = new TestRtpReceiver(); |
| 48 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock)); | 48 receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock)); |
| 49 RtpRtcp::Configuration configuration; | 49 RtpRtcp::Configuration configuration; |
| 50 configuration.audio = false; | 50 configuration.audio = false; |
| 51 configuration.clock = &fake_clock; | 51 configuration.clock = &fake_clock; |
| 52 configuration.outgoing_transport = transport_; | 52 configuration.outgoing_transport = transport_; |
| 53 configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; | 53 configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; |
| 54 | 54 |
| 55 video_module_ = RtpRtcp::CreateRtpRtcp(configuration); | 55 video_module_ = RtpRtcp::CreateRtpRtcp(configuration); |
| 56 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( | 56 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( |
| 57 &fake_clock, receiver_, NULL, &rtp_payload_registry_)); | 57 &fake_clock, receiver_, nullptr, &rtp_payload_registry_)); |
| 58 | 58 |
| 59 video_module_->SetRTCPStatus(RtcpMode::kCompound); | 59 video_module_->SetRTCPStatus(RtcpMode::kCompound); |
| 60 video_module_->SetSSRC(test_ssrc_); | 60 video_module_->SetSSRC(test_ssrc_); |
| 61 video_module_->SetStorePacketsStatus(true, 600); | 61 video_module_->SetStorePacketsStatus(true, 600); |
| 62 EXPECT_EQ(0, video_module_->SetSendingStatus(true)); | 62 EXPECT_EQ(0, video_module_->SetSendingStatus(true)); |
| 63 | 63 |
| 64 transport_->SetSendModule(video_module_, &rtp_payload_registry_, | 64 transport_->SetSendModule(video_module_, &rtp_payload_registry_, |
| 65 rtp_receiver_.get(), receive_statistics_.get()); | 65 rtp_receiver_.get(), receive_statistics_.get()); |
| 66 | 66 |
| 67 VideoCodec video_codec; | 67 VideoCodec video_codec; |
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| 175 payload_specific, true)); | 175 payload_specific, true)); |
| 176 EXPECT_EQ(0u, receiver_->payload_size()); | 176 EXPECT_EQ(0u, receiver_->payload_size()); |
| 177 EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength); | 177 EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength); |
| 178 } | 178 } |
| 179 timestamp += 3000; | 179 timestamp += 3000; |
| 180 fake_clock.AdvanceTimeMilliseconds(33); | 180 fake_clock.AdvanceTimeMilliseconds(33); |
| 181 } | 181 } |
| 182 } | 182 } |
| 183 | 183 |
| 184 } // namespace webrtc | 184 } // namespace webrtc |
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